From devel at thom.fr.eu.org  Mon Mar  1 00:10:24 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 01 Mar 2010 09:10:24 +0100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
 specifying fxs/fxo-channels
In-Reply-To: <3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
	<3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
Message-ID: <cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>

On Sun, 28 Feb 2010 12:11:33 +1100, Brian May
<brian at microcomaustralia.com.au> wrote:
> On 24 February 2010 20:25, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
>> I use sangoma card and the openzap file is generated by the Setup
script
>> from sangoma driver.
>> It seems that the terminology used by zaptel is not used in wanpipe
>> configuration.
> 
> Yes, that is correct.
> 
>> I have an A400 card with an FXO module (providing ports 11 and 12) and
an
>> FXS module (providing ports 9 and 10)
>>
>> My openzap.conf is like this :
>>
>> [span wanpipe FXS]
>> name => Analog phone 1
>> number => 9000
>> fxs-channel => 1:9
>> name => Analog phone 2
>> number => 9001
>> fxs-channel => 1:10
>>
>> [span wanpipe FXO]
>> name => POTS line 1
>> number => 1234567890
>> fxo-channel => 1:11
>> name => POTS line 2
>> number => 0987654321
>> fxo-channel => 1:12
> 
> So ports 9 and 10 are actually FXO ports - extension ports; ports 11
> and 12 are FXS ports, or telephone lines. This is what I have been
> saying.
> 
> Oh, wait, no it isn't. Looks like I was confused. :-(
> 
> It matches my config however.
> 
> Hopefully this fixed the problems with the wiki:
> 
>
http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18693&oldid=18491

On that point, that looks correct (maybe it would make it clearer with a
sentence between parenthesis sayin "using FXO/FXS signaling").

> 
> My guess is that this change is needed also (not absolutely sure here):
> 

On this one, I would have to check in the code. I don't know if the span
order here makes a difference (my guess is no).

>
http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18694&oldid=18693


From lukasz at voiceworks.pl  Mon Mar  1 01:32:03 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 01 Mar 2010 10:32:03 +0100
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
	<7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
Message-ID: <4B8B8993.7010002@voiceworks.pl>

First disable display IE in smg_pri.conf

disable_display_ie=yes

then try to set TON and NP (you can check what telco is sending in 
incoming SETUP) for example:

callingparty_dialplan=national
calledparty_dialplan=national

Your configuration for telco port should be more like;

signalling=pri_cpe
switchtype=euroisdn
callingparty_dialplan=national
calledparty_dialplan=national
disable_display_ie=yes
group=1
spans=1

of course caller_id_number should be set (also check if you telco 
require 04439114600 or 4439114600).

Regards,
Lukasz

lakshmanan ganapathy wrote:
> Dear Moy,
> That's didn't seem to solve the problem.
> I gave the following command.
> originate 
> {origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266 
> &bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)
>
> The D-Chan Log is
> http://pastebin.freeswitch.org/12268
>
> Kindly refer the attached pcap file that I captured with wanpipemon 
> utility. I think it might help.
>
>
> On Sun, Feb 28, 2010 at 12:05 AM, Moises Silva <moises.silva at gmail.com 
> <mailto:moises.silva at gmail.com>> wrote:
>
>     I believe the problem FreeSWITCH is setting that as a default
>     callerid name, which your telco does not like.
>
>     Try setting the caller id name and number by yourself as explained
>     in the "originate" section
>     here http://wiki.freeswitch.org/wiki/Mod_commands
>
>
>     On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy
>     <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>         I think it says Invalid Information Element for the DISPLAY
>         smg_prid/a/8122133885!!!
>         correct?? If so, can you please help me to solve this?
>
>
>         On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy
>         <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>             In the Dchan log it is saying Invalid Information
>             Elements. That might be a problem??? But I even don't know
>             why it is saying Invalid Information Element??
>             Please guide me!!!
>
>
>
>             On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy
>             <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>                 Dear Moy,
>                 Here are the details:
>
>                 FreeSwitch Log:
>                 http://pastebin.freeswitch.org/12256
>
>                  /var/log/sangoma_pri/dchan_<span-no>.log:
>                  http://pastebin.freeswitch.org/12257
>
>                 /var/log/sangoma_mgd.log:
>                 http://pastebin.freeswitch.org/12258
>
>                 smg_pri.conf
>                 http://pastebin.freeswitch.org/12259
>
>
>
>                 On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva
>                 <moises.silva at gmail.com
>                 <mailto:moises.silva at gmail.com>> wrote:
>
>                     Hello lakshmanan,
>
>                     Please enable the dchan_log=q931 in
>                     /etc/wanpipe/smg_pri.conf and then restart it
>                     (smg_ctrl restart), then pastebin the logs
>
>                      /var/log/sangoma_pri/dchan_<span-no>.log
>                     /var/log/sangoma_mgd.log
>
>                     That will contain the Q931 details (if any). Also
>                     pastebin your smg_pri.conf.
>
>                     Also enable the FreeSWITCH debug logging (see the
>                     FreeSWITCH wiki for details about that) and paste
>                     them too.
>
>                     -- 
>                     Moises Silva
>                     Senior Software Engineer
>                     Sangoma Technologies Inc. | 50 McIntosh Drive,
>                     Suite 120, Markham ON L3R 9T3 Canada
>                     t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>                     <mailto:moy at sangoma.com>
>
>                     On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan
>                     ganapathy <lakindia89 at gmail.com
>                     <mailto:lakindia89 at gmail.com>> wrote:
>
>                         Dear all,
>                         I'm having a A102 Sangoma hardware. I
>                         configured it with freeswitch.
>                         wanrouter status, says both the port as connected.
>                         My smg_prid version is
>
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
>                         ================System restart=============
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Sangoma PRI Protocol Stack Daemon       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Version: 1.54                           =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Date: Feb 15 2010                       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Wanpipe Release: wanpipe-3.5.8.6       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Revision:Revision: 15288                =
>                         Feb 26 16:08:14 FMS-FreeSwitch
>                         sangoma_prid:                              
>                         ===========================================
>
>                         My freeswitch version is 16729.
>                         I started freeswitch.
>
>                         oz list
>                         +OK
>                         span: 1 (smg_prid)
>                         type: Sangoma (boost)
>                         chan_count: 60
>                         dialplan: XML
>                         context: default
>                         dial_regex:
>                         fail_dial_regex:
>                         hold_music:
>                         analog_options none
>
>                         I originated a call as
>                         originate openzap/smg_prid/a/9952248266
>                         &park(), which hits my mobile.
>
>                         But when I issued the following command:
>                         originate openzap/smg_prid/a/9952248266
>                         &bridge(openzap/smg_prid/a/8122133885)
>                         It rings my mobile (9952248266) first, but
>                         after that the following error was displayed
>
>                         2010-02-26 16:20:51.736080 [ERR]
>                         switch_ivr_originate.c:2387 Cannot create
>                         outgoing channel of type [openzap] cause:
>                         [NORMAL_CIRCUIT_CONGESTION]
>                         The call got ended in my mobile.
>
>                         Freeswitch log and smg_pri.conf
>                                   http://pastebin.freeswitch.org/12248
>                         openzap.conf:
>                         [span wanpipe smg_prid]
>                         name => smg_prid
>                         trunk_type =>e1
>                         b-channel => 1:1-15
>                         b-channel => 1:17-31
>                         trunk_type =>e1
>                         b-channel => 2:1-15
>                         b-channel => 2:17-31
>
>                         openzap.conf.xml:
>                         <configuration name="openzap.conf"
>                         description="OpenZAP Configuration">
>                           <settings>
>                             <param name="debug" value="0"/>
>                           </settings>
>                           <boost_spans>
>                                 <span id="smg_prid">
>                                 <param name="dialplan" value="XML"/>
>                                 <param name="context" value="default"/>
>                                 </span>
>                           </boost_spans>
>                         </configuration>
>
>                         Please guide me to setup this one!!.
>
>
>                         _______________________________________________
>                         FreeSWITCH-users mailing list
>                         FreeSWITCH-users at lists.freeswitch.org
>                         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                         http://www.freeswitch.org
>
>
>
>                     _______________________________________________
>                     FreeSWITCH-users mailing list
>                     FreeSWITCH-users at lists.freeswitch.org
>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                     http://www.freeswitch.org
>
>
>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     -- 
>     Moises Silva
>     Senior Software Engineer
>     Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham
>     ON L3R 9T3 Canada
>     t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From michal.kalinowski at interia.pl  Mon Mar  1 03:51:55 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 1 Mar 2010 12:51:55 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
Message-ID: <7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>

Thank's Brian

So at this moment i have only clean configuration in lua without any XML.

I get example from wiki:

hash={["main"]=undef,

   ["name"]="top",

         ["greet_long"]="phrase:demo_ivr_main_menu",

["greet_short"]="phrase:demo_ivr_main_menu_short",

["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",

["exit_sound"]="voicemail/vmgoodbye.wav",

["confirm_macro"]="undef",

 ["confirm_key"]="undef",

    ["confirm_attempts"]="3",

      ["inter_digit_timeout"]="2000",

       ["digit_len"]="1",

               ["timeout"]="10000",
["max_failures"]="3"
}
top = freeswitch.IVRMenu(hash["main"],
hash["name"],
hash["greet_long"],hash["greet_short"],
hash["invalid_sound"],
hash["exit_sound"],
hash["confirm_macro"],
hash["confirm_key"],
hash["confirm_attempts"],
hash["inter_digit_timeout"],
hash["digit_len"],
hash["timeout"],
hash["max_failures"]);

top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
top:execute(session, "top");

But something is wrong freeswitch say
2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
expected 16..16 args, got 13
stack traceback:
        [C]: in function 'IVRMenu'
        /usr/local/freeswitch/scripts/ivr.lua:19: in main chunk

What args FS expect?

BR,
Micha?
W dniu 27 lutego 2010 22:48 u?ytkownik Brian West
<brian at freeswitch.org> napisa?:
> can you point out on the wiki that indicated you are able to dot his? ?You're mixing two concepts incorrectly here.
>
> For example you can't do "set" and exec the file to get the contents.. you can however on linux use "exec" instead of set. ?But your script needs to print it to stdout. ?Your example in your lua script is for the config engine in in lua thats like XML curl... in which case you're not building the full document like you should. ?Read thru the XML CURL docs if you want to do "XML_STRING = mydialplan".
>
> /b
>
>
> On Feb 27, 2010, at 3:10 PM, michal kalinowski wrote:
>
>> Coming back to this case I create in lua some script with XML ivr.
>>
>> #!/usr/local/bin/lua
>>
>> mydialplan = [[
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <include>
>> ?<!-- demo IVR setup -->
>> ?<!-- demo IVR, Main Menu -->
>> ?<menu name="demo_ivr"
>> ? ? ?greet-long="phrase:demo_ivr_main_menu"
>> ? ? ?greet-short="phrase:demo_ivr_main_menu_short"
>> ? ? ?invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>> ? ? ?exit-sound="voicemail/vm-goodbye.wav"
>> ? ? ?confirm-macro=""
>> ? ? ?confirm-key=""
>> ? ? ?tts-engine="flite"
>> ? ? ?tts-voice="rms"
>> ? ? ?confirm-attempts="3"
>> ? ? ?timeout="10000"
>> ? ? ?inter-digit-timeout="2000"
>> ? ? ?max-failures="3"
>> ? ? ?max-timeouts="3"
>> ? ? ?digit-len="4">
>>
>> ? ?<!-- The following are the definitions for the digits the user dials -->
>> ? ?<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>> ? ?<entry action="menu-exec-app" digits="1" param="bridge
>> sofia/$${domain}/888 at conference.freeswitch.org"/>
>> ? ?<entry action="menu-exec-app" digits="2" param="transfer 9996 XML
>> default"/> ? ?<!-- FS echo -->
>> ? ?<entry action="menu-exec-app" digits="3" param="transfer 9999 XML
>> default"/> ? ?<!-- MOH -->
>> ? ?<entry action="menu-exec-app" digits="4" param="transfer 9991 XML
>> default"/> ? ?<!-- ClueCon -->
>> ? ?<entry action="menu-exec-app" digits="5" param="transfer 1234*256
>> enum"/> ? ? ? <!-- Screaming monkeys -->
>> ? ?<entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
>> ? ? ? ? ? ? <!-- demo sub menu -->
>> ? ?<!-- Using a regex in the digits tag lets you define a dial
>> pattern for the caller
>> ? ? ? ? You may define multiple regexes if you need a different
>> pattern for some reason -->
>> ? ?<entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
>> param="transfer $1 XML features"/>
>> ? ?<entry action="menu-top" digits="9"/> ? ? ? ? ?<!-- Repeat this menu -->
>> ?</menu>
>>
>> </include>
>> ]]
>> XML_STRING = mydialplan
>>
>> in dialplan I have context with this ivr
>>
>> ?<context name="ivr">
>> <extension name="ivr">
>> ? ? <condition field="caller_id_number" expression="^[0-9]"/>
>> ? ? <condition field="destination_number" expression="^(\d{11})$">
>> ? ? ? ?<action application="answer"/>
>> ? ? ? ?<action application="ivr" data="demo_ivr"/>
>> ? ? </condition>
>> </extension>
>> ?</context>
>>
>> in ivr.conf i have this
>>
>> <configuration name="ivr.conf" description="IVR menus">
>> ?<menus>
>> ? ? ? ?<X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
>> ?</menus>
>> </configuration>
>>
>> but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
>> mod_dptools.c:1247 Unable to find menu"
>> what I do wrong ?
>>
>>
>> BR,
>> Micha?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From moises.silva at gmail.com  Mon Mar  1 06:55:57 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 1 Mar 2010 09:55:57 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
	<7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
Message-ID: <c4d05cbe1003010655v706a970cxf999c26c3a32a7a0@mail.gmail.com>

Hi again,

On Mon, Mar 1, 2010 at 1:16 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear Moy,
> That's didn't seem to solve the problem.
> I gave the following command.
> originate
> {origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266
> &bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)
>

The engineer in charge of the PRI daemon will be sending you an e-mail soon
to fix your problem. Let me know if you don't get the e-mail today.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From mmg at transtelco.net  Mon Mar  1 08:09:05 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 11:09:05 -0500
Subject: [Freeswitch-users] High CPU usage on 1.0.5
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555B40@VMBX113.ihostexchange.net>

Dear freeswitch group

I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU usage even if there are only a few calls on the system or no calls at all. We are running Debian with kernel 2.6.26-2-686

Anyone experimenting a similar issue?


Version
freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100225-0400 (16810M)


Thanks in advance




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From sos at sokhapkin.dyndns.org  Mon Mar  1 08:32:32 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Mar 2010 11:32:32 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
Message-ID: <201003011132.32415.sos@sokhapkin.dyndns.org>

How much is "high" and on which CPU?

On Monday 01 March 2010, Manuel Mar?n wrote:
> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at all.
> We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
>
> Thanks in advance
>
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
>
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214




From brian at freeswitch.org  Mon Mar  1 08:37:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 10:37:27 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <201003011132.32415.sos@sokhapkin.dyndns.org>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
Message-ID: <FB95AD29-58EC-4C98-B744-9E7606EC489C@freeswitch.org>

Please type uname -a and give me the contents of /proc/cpuinfo

/b

On Mar 1, 2010, at 10:32 AM, Sergey Okhapkin wrote:

> How much is "high" and on which CPU?



From mmg at transtelco.net  Mon Mar  1 08:41:26 2010
From: mmg at transtelco.net (=?utf-8?B?TWFudWVsIE1hcsOtbg==?=)
Date: Mon, 1 Mar 2010 11:41:26 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <201003011132.32415.sos@sokhapkin.dyndns.org>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>

Hi Sergey

Attached you will find a screenshot of the CPU usage. There were only 22 calls running. Is it normal on release 1.0.5?


processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 0
initial apicid  : 0
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7188.28
clflush size    : 64
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 3
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 6
initial apicid  : 6
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.45
clflush size    : 64
power management:

processor       : 2
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 1
initial apicid  : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.41
clflush size    : 64
power management:

processor       : 3
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 3
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 7
initial apicid  : 7
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.46
clflush size    : 64
power management:







Manuel Mar?n 
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214


-----Original Message-----
From: Sergey Okhapkin [mailto:sos at sokhapkin.dyndns.org] 
Sent: Monday, March 01, 2010 9:33 AM
To: freeswitch-users at lists.freeswitch.org
Cc: Manuel Mar?n
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5

How much is "high" and on which CPU?

On Monday 01 March 2010, Manuel Mar?n wrote:
> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at all.
> We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
>
> Thanks in advance
>
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
>
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214


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From anthony.minessale at gmail.com  Mon Mar  1 08:47:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 10:47:58 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
Message-ID: <191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>

we are still wary about 5.3 due to bugs reported in libc.
What number does it say it detected for the gap?


On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:

> OK - I've realised I do get the same warning with CentOS 5.3, it just goes
> past more quickly so I didn't see it.  Maybe it is just the hardware ....
>
>
> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi,
>>
>> I wonder if anyone would be able to advise please:
>>
>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
>> start FreeSWITCH that
>>
>> "Abnormally large timer gap detected"
>> "Do you have your kernel timer set to greater than 1kHz?  You may
>> experience audio problems".
>>
>> I get no such warning if I build on CentOS 5.3, and the test timings it
>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>> to Centos5.4 was a yum update on the 5.3 build.
>>
>> I guess the warning comes from here:
>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>
>> This is all on pretty low spec hardware - a couple of different Dell
>> optiplex p4's I use for testing.
>>
>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>> 5.4 and not worry about the warnings, or if there is something I can do to
>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>> use such crummy hardware?!
>>
>> Many thanks in advance
>> Bruce
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tomb at cachecomm.com  Mon Mar  1 08:42:39 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 09:42:39 -0700
Subject: [Freeswitch-users] NAT issue?
Message-ID: <4B8BEE7F.1030401@cachecomm.com>

Voip Gods,

I am a newbie to freeswitch i have a problem with nat i think . i can 
only get one sip phone to register with freeswitch here is my setup
FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only 
register phone one . when freeswitch loads it says set ext-ip to my 
gateway IP .  i know  it something super stupid  can some one give me a 
boot in the right direction

Cheers
 Tom



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



From brian at freeswitch.org  Mon Mar  1 08:50:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 10:50:36 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
	<4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>
Message-ID: <A82574D6-9274-4D61-8E14-D5BAD18C8B85@freeswitch.org>

Please output of uname -a?

/b

On Mar 1, 2010, at 10:41 AM, Manuel Mar?n wrote:

> Attached you will find a screenshot of the CPU usage. There were only 22 calls running. Is it normal on release 1.0.5?



From mmg at transtelco.net  Mon Mar  1 08:53:10 2010
From: mmg at transtelco.net (=?utf-8?B?TWFudWVsIE1hcsOtbg==?=)
Date: Mon, 1 Mar 2010 08:53:10 -0800
Subject: [Freeswitch-users] High CPU usage 1.0.5
Message-ID: <20100301085310.B5C9FD1D@resin15.mta.everyone.net>


Brian

Attached you will find the uname -a output

rtpb2:/var/log# uname -a
Linux rtpb2.transtelco.net 2.6.26-2-686 #1 SMP Wed Feb 10 08:59:21 UTC 2010 i686 GNU/Linux



--- brian at freeswitch.org wrote:

From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5
Date: Mon, 1 Mar 2010 10:37:27 -0600

Please type uname -a and give me the contents of /proc/cpuinfo

/b

On Mar 1, 2010, at 10:32 AM, Sergey Okhapkin wrote:

> How much is "high" and on which CPU?


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




From christian.loeschenkohl at xpirio.com  Mon Mar  1 09:08:17 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 01 Mar 2010 18:08:17 +0100
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
Message-ID: <4B8BF481.5000400@xpirio.com>

hello

have you tried the -vm startup flag?

anthony stated on an similar question (from me)
---
If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that may not manifest into the load average.
---

i also use a new trunk version, after using -vm the load where normal again

br


On 2010-03-01 06:20, Manuel Mar?n wrote:

> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at
> all. We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
> Thanks in advance
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
> Manuel Mar?n
>
> Transtelco
>
> US 1.915.2172232
>
> MX 52.656.6921109
>
> FAX 1.915.2311214
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From msc at freeswitch.org  Mon Mar  1 09:24:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 09:24:56 -0800
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8BEE7F.1030401@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
Message-ID: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>

On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com> wrote:

> Voip Gods,
>
> I am a newbie to freeswitch i have a problem with nat i think . i can
> only get one sip phone to register with freeswitch here is my setup
> FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
> register phone one . when freeswitch loads it says set ext-ip to my
> gateway IP .  i know  it something super stupid  can some one give me a
> boot in the right direction
>
> Cheers
>  Tom
>
> What device is performing the NAT? Make sure it doesn't have an ALG in
there as well. See this new page that we have been updating:
http://wiki.freeswitch.org/wiki/ALG

Also, which phones are you using?
-MC
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From mmg at transtelco.net  Mon Mar  1 09:25:08 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 12:25:08 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4B8BF481.5000400@xpirio.com>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<4B8BF481.5000400@xpirio.com>
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>

Thanks Christian. I'll try with the -vm startup flag. I think that issue is related to timing because even if there are no calls running on the system the CPU load is 50-70%




Manuel Mar?n
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214


-----Original Message-----
From: Christian L?schenkohl [mailto:christian.loeschenkohl at xpirio.com]
Sent: Monday, March 01, 2010 10:08 AM
To: freeswitch-users at lists.freeswitch.org
Cc: Manuel Mar?n
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5

hello

have you tried the -vm startup flag?

anthony stated on an similar question (from me)
---
If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that may not manifest into the load average.
---

i also use a new trunk version, after using -vm the load where normal again

br


On 2010-03-01 06:20, Manuel Mar?n wrote:

> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at
> all. We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
> Thanks in advance
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
> Manuel Mar?n
>
> Transtelco
>
> US 1.915.2172232
>
> MX 52.656.6921109
>
> FAX 1.915.2311214
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


--
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From codecomplete at free.fr  Mon Mar  1 10:08:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Mar 2010 19:08:00 +0100
Subject: [Freeswitch-users] [init.d script] Possible improvement?
Message-ID: <ve0oo5hcev0ij4hbjkpg8liju4ua7159fp@4ax.com>

Hello

I use the sample Init script in the wiki to have Freeswitch start
automatically at boot-time on my test CentOS 5.4 host, but I notice
that it takes several seconds for FS to be up and running before I can
connect to it with fs_cli:

========
# /etc/init.d/freeswitch restart
Shutting down freeswitch: [  OK  ]
Starting freeswitch: [  OK  ]

# fs_cli
[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket
Connection Error]
========

I don't know much about Init scripts: Is there a way to display some
information so I know when the script is done and FS is ready accept
commands from the CLI?

Thank you.



From anthony.minessale at gmail.com  Mon Mar  1 10:38:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 12:38:59 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
Message-ID: <191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>

I am trying my best not to be completely annoyed at the "somewhat limiting"
remark after going over the countless hours I spent making sure this feature
work and then being patient with 2 people insisting there was a bug when
they both shared an identical misconfiguration.

If we started doing DNS lookups everywhere so hostname could be interchanged
with IP we would have large blocking all over the place when the DNS was not
available and horribly messy code doing all the lookups.  There was already
an experiment called project asterisk that proved the flaws in this design
approach.

FreeSWITCH requires that you keep the hostname uniform for all configuration
be it a domain name or an IP so long as its identical everywhere.  an IP
*is* a domain in sip it's not another form of some DNS lookup.


On Sun, Feb 28, 2010 at 12:23 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> This is one of the most important feature my users want. They don't care
> how I do it, they are just happy it works.
>
>                         Thanks! __Yehavi:
>
> 2010/2/28 Brian West <brian at freeswitch.org>
>
> Really?  Come on guys... the feature is something you can't find elsewhere
>> without paying and you're all still not totally pleased with it?
>>
>> /me shakes his head.
>>
>> We have already started talking about how to make the feature more robust.
>>
>> /b
>>
>> On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
>>
>> > Hello Gabe,
>> >
>> >   I agree that this is somewhat limiting, but with Polycom's central
>> provisioning (via XML files) I don't see this as a major drawback.
>> >
>> >                          __Yehavi:
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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pstn:+19193869900
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From msc at freeswitch.org  Mon Mar  1 11:09:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 11:09:30 -0800
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
	<7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
Message-ID: <87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>

2010/3/1 michal kalinowski <michal.kalinowski at interia.pl>

> Thank's Brian
>
> So at this moment i have only clean configuration in lua without any XML.
>
> I get example from wiki:
>
> hash={["main"]=undef,
>
>   ["name"]="top",
>
>         ["greet_long"]="phrase:demo_ivr_main_menu",
>
> ["greet_short"]="phrase:demo_ivr_main_menu_short",
>
> ["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",
>
> ["exit_sound"]="voicemail/vmgoodbye.wav",
>
> ["confirm_macro"]="undef",
>
>  ["confirm_key"]="undef",
>
>    ["confirm_attempts"]="3",
>
>      ["inter_digit_timeout"]="2000",
>
>       ["digit_len"]="1",
>
>               ["timeout"]="10000",
> ["max_failures"]="3"
> }
> top = freeswitch.IVRMenu(hash["main"],
> hash["name"],
> hash["greet_long"],hash["greet_short"],
> hash["invalid_sound"],
> hash["exit_sound"],
> hash["confirm_macro"],
> hash["confirm_key"],
> hash["confirm_attempts"],
> hash["inter_digit_timeout"],
> hash["digit_len"],
> hash["timeout"],
> hash["max_failures"]);
>
> top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
> top:execute(session, "top");
>
> But something is wrong freeswitch say
> 2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
> expected 16..16 args, got 13
> stack traceback:
>        [C]: in function 'IVRMenu'
>        /usr/local/freeswitch/scripts/ivr.lua:19: in main chunk
>
> What args FS expect?
>

I ran into this issue when I was converting the demo IVR to Lua. There are
indeed 16 arguments but you've only got 13 of them. Here's a snippet from
the Lua script I'm working on:
ivr_def = {
    ["main"]                = undef,
    ["name"]                = "demo_ivr_lua",
    ["greet_long"]          = "phrase:demo_ivr_main_menu",
    ["greet_short"]         = "phrase:demo_ivr_main_menu_short",
    ["invalid_sound"]       = "ivr/ivr-that_was_an_invalid_entry.wav",
    ["exit_sound"]          = "voicemail/vm-goodbye.wav",
    ["confirm_macro"]       = "",
    ["confirm_key"]         = "",
    ["tts_engine"]          = "flite",
    ["tts_voice"]           = "rms",
    ["confirm_attempts"]    = "3",
    ["inter_digit_timeout"] = "2000",
    ["digit_len"]           = "4",
    ["timeout"]             = "10000",
    ["max_failures"]        = "3",
    ["max_timeouts"]        = "2"
}
Those are all 16 arguments. Notice you need tts_engine, tts_voice, and
max_timeouts. Try that and let us know how it goes.

-MC
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From anthony.minessale at gmail.com  Mon Mar  1 11:29:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:29:37 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
Message-ID: <191c3a031003011129q777468d0ve852e3ad7363dd3a@mail.gmail.com>

Lesson 1,

NAT Issue is a redundant statement like Failure Problem



On Mon, Mar 1, 2010 at 11:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com> wrote:
>
>> Voip Gods,
>>
>> I am a newbie to freeswitch i have a problem with nat i think . i can
>> only get one sip phone to register with freeswitch here is my setup
>> FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
>> register phone one . when freeswitch loads it says set ext-ip to my
>> gateway IP .  i know  it something super stupid  can some one give me a
>> boot in the right direction
>>
>> Cheers
>>  Tom
>>
>> What device is performing the NAT? Make sure it doesn't have an ALG in
> there as well. See this new page that we have been updating:
> http://wiki.freeswitch.org/wiki/ALG
>
> Also, which phones are you using?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Mon Mar  1 11:30:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 1 Mar 2010 21:30:26 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
Message-ID: <d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>

Hello Anthony,


 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>

> I am trying my best not to be completely annoyed at the "somewhat limiting"
> remark after going over the countless hours I spent making sure this feature
> work and then being patient with 2 people insisting there was a bug when
> they both shared an identical misconfiguration.
>
"somewhat limiting" was because my experiments up to now with Polycom and
SIP/TLS were successfull with domain names only. However, I am going to
inverstigate this further to see how this can work with IP only.

In the meantime I've documented the required configuration under the Polycom
wiki page.


> If we started doing DNS lookups everywhere so hostname could be
> interchanged with IP we would have large blocking all over the place when
> the DNS was not available and horribly messy code doing all the lookups.
> There was already an experiment called project asterisk that proved the
> flaws in this design approach.
>
> I think that's why some of us move from Asterisk to FreeSwitch. I also
preffer less to no DNS lookups on a critical system which must work even
when there are some kinds of communication problems.

                      Thanks, __Yehavi:
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From anthony.minessale at gmail.com  Mon Mar  1 11:31:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:31:46 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<4B8BF481.5000400@xpirio.com>
	<4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>
Message-ID: <191c3a031003011131q5ea99122w1b987ee0d94979a3@mail.gmail.com>

update to SVN trunk where that option is now default.

If you are getting a lot of cpu usage on an idle box,
update both FS and upgrade or downgrade your kernel away from the "tickless
bug"


On Mon, Mar 1, 2010 at 11:25 AM, Manuel Mar?n <mmg at transtelco.net> wrote:

> Thanks Christian. I'll try with the -vm startup flag. I think that issue is
> related to timing because even if there are no calls running on the system
> the CPU load is 50-70%
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214
>
>
> -----Original Message-----
> From: Christian L?schenkohl [mailto:christian.loeschenkohl at xpirio.com]
> Sent: Monday, March 01, 2010 10:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Cc: Manuel Mar?n
> Subject: Re: [Freeswitch-users] High CPU usage 1.0.5
>
> hello
>
> have you tried the -vm startup flag?
>
> anthony stated on an similar question (from me)
> ---
> If you have a box that has trouble with timing it could cost more
> resources.
> you can always run freeswitch -vm to use an alternate form of timing that
> may not manifest into the load average.
> ---
>
> i also use a new trunk version, after using -vm the load where normal again
>
> br
>
>
> On 2010-03-01 06:20, Manuel Mar?n wrote:
>
> > Dear freeswitch group
> >
> > I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> > usage even if there are only a few calls on the system or no calls at
> > all. We are running Debian with kernel 2.6.26-2-686
> >
> > Anyone experimenting a similar issue?
> >
> > Thanks in advance
> >
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
> >
> > Manuel Mar?n
> >
> > Transtelco
> >
> > US 1.915.2172232
> >
> > MX 52.656.6921109
> >
> > FAX 1.915.2311214
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar  1 11:42:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:42:39 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
Message-ID: <191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>

you can use names if you use *all names*
you just can't mix, but all IP is much faster and safer because there is no
need to do constant dns lookups on something that never changes anyway, you
can easily alias your profile to a friendly name, it also makes it more
portable.




On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello Anthony,
>
>
>  2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>
>> I am trying my best not to be completely annoyed at the "somewhat
>> limiting" remark after going over the countless hours I spent making sure
>> this feature work and then being patient with 2 people insisting there was a
>> bug when they both shared an identical misconfiguration.
>>
> "somewhat limiting" was because my experiments up to now with Polycom and
> SIP/TLS were successfull with domain names only. However, I am going to
> inverstigate this further to see how this can work with IP only.
>
> In the meantime I've documented the required configuration under the
> Polycom wiki page.
>
>
>> If we started doing DNS lookups everywhere so hostname could be
>> interchanged with IP we would have large blocking all over the place when
>> the DNS was not available and horribly messy code doing all the lookups.
>> There was already an experiment called project asterisk that proved the
>> flaws in this design approach.
>>
>> I think that's why some of us move from Asterisk to FreeSwitch. I also
> preffer less to no DNS lookups on a critical system which must work even
> when there are some kinds of communication problems.
>
>                       Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tomb at cachecomm.com  Mon Mar  1 12:44:44 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 13:44:44 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
Message-ID: <4B8C273C.6010808@cachecomm.com>

Michael Collins wrote:
>
>
> On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com 
> <mailto:tomb at cachecomm.com>> wrote:
>
>     Voip Gods,
>
>     I am a newbie to freeswitch i have a problem with nat i think . i can
>     only get one sip phone to register with freeswitch here is my setup
>     FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
>     register phone one . when freeswitch loads it says set ext-ip to my
>     gateway IP .  i know  it something super stupid  can some one give
>     me a
>     boot in the right direction
>
>     Cheers
>      Tom
>
> What device is performing the NAT? Make sure it doesn't have an ALG in 
> there as well. See this new page that we have been updating:
> http://wiki.freeswitch.org/wiki/ALG
>
> Also, which phones are you using?
> -MC
>
>
>
MC,
the phone are both Aastra they are behind a netgear fvs338 which from 
your post are very bad.  I will try to disable the ALG if i can and try 
again.

Thanks for the help

Cheers
Tom


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



From brian at freeswitch.org  Mon Mar  1 12:51:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 14:51:22 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8C273C.6010808@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
Message-ID: <FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>

The only disable button on a netgear is to a sledge hammer and go buy something that doesn't mess with your SIP traffic  :P

/b

On Mar 1, 2010, at 2:44 PM, Tom wrote:

> MC,
> the phone are both Aastra they are behind a netgear fvs338 which from 
> your post are very bad.  I will try to disable the ALG if i can and try 
> again.
> 
> Thanks for the help
> 
> Cheers
> Tom



From gkuri at ieee.org  Mon Mar  1 13:09:39 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 1 Mar 2010 13:09:39 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
Message-ID: <8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>

Hi Anthony,

I appreciate all the hard work yourself, Brian, Mike and all the other
developers have put in FS! By no means am I discounting any of it,
it's truly invaluable.

Our "philosophy" has always been to use DNS where available, so that
we're not worrying about re-configuring a bunch of devices if IP
addresses change and this obviously goes to setting the phones up as
well to use DNS for the proxy and registrar fields. And it allows us
to do fancy things, like split-view DNS, where if a phone is taken
off-site and we want it to remotely register to FS, it will get the
public IP address of the FS server, rather than the private, without
re-configuring the phone.

If I'm understanding your email correctly, you're saying we need to
use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
DNS, but perhaps I'm missing something somewhere. How would we be able
to use the DNS name of our SIP domain in phones (rather than IP) and
ensure SLA works for both incoming/outgoing calls? Is that possible at
this point in time or do we need to use the IP addresses in the proxy
field on the phone if we want to use SLA?

Thanks for all your help.

Gabriel Kuri


On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you can use names if you use *all names*
> you just can't mix, but all IP is much faster and safer because there is no
> need to do constant dns lookups on something that never changes anyway, you
> can easily alias your profile to a friendly name, it also makes it more
> portable.
>
>
>
>
> On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>>
>> Hello Anthony,
>>
>> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> I am trying my best not to be completely annoyed at the "somewhat
>>> limiting" remark after going over the countless hours I spent making sure
>>> this feature work and then being patient with 2 people insisting there was a
>>> bug when they both shared an identical misconfiguration.
>>
>> "somewhat limiting" was because my experiments up to now with Polycom and
>> SIP/TLS were successfull with domain names only. However, I am going to
>> inverstigate this further to see how this can work with IP only.
>>
>> In the meantime I've documented the required configuration under the
>> Polycom wiki page.
>>
>>>
>>> If we started doing DNS lookups everywhere so hostname could be
>>> interchanged with IP we would have large blocking all over the place when
>>> the DNS was not available and horribly messy code doing all the lookups.
>>> There was already an experiment called project asterisk that proved the
>>> flaws in this design approach.
>>>
>> I think that's why some of us move from Asterisk to FreeSwitch. I also
>> preffer less to no DNS lookups on a critical system which must?work even
>> when there are some kinds of communication problems.
>>
>> ????????????????????? Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Mon Mar  1 13:22:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 15:22:55 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
Message-ID: <191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>

I really want to be done with this thread......

You were using the IP in some places and the hostnames in others, this makes
it so the SQL statements not match because sql statements cannot do dns
lookups.  I will not be doing any work to resolve hostnames back and forth
to IP inside FS.


The subscribes were using the hostname and the db and invites were using the
IP.

you have to have the same exact domain in your user directory and have the
force-register-domain and force-register-domain-db also set to that value or
omit them completely and use the domain names in 100% of the config fields
on the phone.




On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Hi Anthony,
>
> I appreciate all the hard work yourself, Brian, Mike and all the other
> developers have put in FS! By no means am I discounting any of it,
> it's truly invaluable.
>
> Our "philosophy" has always been to use DNS where available, so that
> we're not worrying about re-configuring a bunch of devices if IP
> addresses change and this obviously goes to setting the phones up as
> well to use DNS for the proxy and registrar fields. And it allows us
> to do fancy things, like split-view DNS, where if a phone is taken
> off-site and we want it to remotely register to FS, it will get the
> public IP address of the FS server, rather than the private, without
> re-configuring the phone.
>
> If I'm understanding your email correctly, you're saying we need to
> use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
> DNS, but perhaps I'm missing something somewhere. How would we be able
> to use the DNS name of our SIP domain in phones (rather than IP) and
> ensure SLA works for both incoming/outgoing calls? Is that possible at
> this point in time or do we need to use the IP addresses in the proxy
> field on the phone if we want to use SLA?
>
> Thanks for all your help.
>
> Gabriel Kuri
>
>
> On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > you can use names if you use *all names*
> > you just can't mix, but all IP is much faster and safer because there is
> no
> > need to do constant dns lookups on something that never changes anyway,
> you
> > can easily alias your profile to a friendly name, it also makes it more
> > portable.
> >
> >
> >
> >
> > On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com>
> > wrote:
> >>
> >> Hello Anthony,
> >>
> >> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
> >>>
> >>> I am trying my best not to be completely annoyed at the "somewhat
> >>> limiting" remark after going over the countless hours I spent making
> sure
> >>> this feature work and then being patient with 2 people insisting there
> was a
> >>> bug when they both shared an identical misconfiguration.
> >>
> >> "somewhat limiting" was because my experiments up to now with Polycom
> and
> >> SIP/TLS were successfull with domain names only. However, I am going to
> >> inverstigate this further to see how this can work with IP only.
> >>
> >> In the meantime I've documented the required configuration under the
> >> Polycom wiki page.
> >>
> >>>
> >>> If we started doing DNS lookups everywhere so hostname could be
> >>> interchanged with IP we would have large blocking all over the place
> when
> >>> the DNS was not available and horribly messy code doing all the
> lookups.
> >>> There was already an experiment called project asterisk that proved the
> >>> flaws in this design approach.
> >>>
> >> I think that's why some of us move from Asterisk to FreeSwitch. I also
> >> preffer less to no DNS lookups on a critical system which must work even
> >> when there are some kinds of communication problems.
> >>
> >>                       Thanks, __Yehavi:
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From wchao at yahoo.com  Mon Mar  1 13:31:29 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Mon, 1 Mar 2010 16:31:29 -0500 (EST)
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
Message-ID: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>

I am using the xml_rpc interface like so:

http://192.168.1.1:8080/webapi/show?calls

I get a response in a table, which is a little bit undesirable because I 
want to parse the result in a program and don't want to parse a bunch of 
HTML (just seems inelegant and wasteful), so I instead tried the following 
request:

http://192.168.1.1:8080/txtapi/show?calls

That works better, but then another wrinkle appears. Caller ID (and Callee 
ID) names sometimes have commas, which messes up the CSV.

This appears to be a bug in mod_commands.c (or I suppose you could call it 
an artifact of somebody coding up something quickly and not worrying about 
the absolute correctness of something that is not critical to the core). 
There is also another bug with show calls via webapi in that the generated 
HTML has spurious <td> tags. I think the spurious <td> tag bug arises due 
to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally, 
another bug (or maybe just unimplemented feature) is that xmlapl isn't 
really different from webapi.

These problems occur in the latest source tree (20100301).

I am wondering if (a) one of the developers can fix these or (b) if I fix 
the bugs, how can I submit patches [and if I submit patches, will they be 
accepted into the main tree]?

The fixes are pretty trivial and I'd be happy to code them up if somebody 
will tell me how I can submit the patches (haven't done it before).


From m.sobkow at marketelsystems.com  Mon Mar  1 13:34:15 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 01 Mar 2010 15:34:15 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
Message-ID: <4B8C32D7.40202@marketelsystems.com>

I'm using the originate command from Erlang, specifying a timeout of 30 
seconds.  It is properly giving up on dialing, but it's still returning 
a call UUID instead of an error/timed out status.

How can I query Freeswitch to determine whether the call was answered or 
timed out, without resorting to the event interface?  (I could put 
together a dialplan that calls the event interface after doing the 
originate, but from what I've seen that event interface won't execute if 
the previous command "fails", such as when I did a play_and_get_digits 
timing out.)

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



From brian at freeswitch.org  Mon Mar  1 13:48:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 15:48:45 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <777F5906-C26D-4D22-B238-426CB833B88A@freeswitch.org>

Just off the top of my head have you tried /xmlapi/? or read the wiki page on this?  http://wiki.freeswitch.org/wiki/Mod_xml_rpc

and "show calls as xml"?

/b


On Mar 1, 2010, at 3:31 PM, Wellie Chao wrote:

> I am using the xml_rpc interface like so:
> 
> http://192.168.1.1:8080/webapi/show?calls
> 
> I get a response in a table, which is a little bit undesirable because I 
> want to parse the result in a program and don't want to parse a bunch of 
> HTML (just seems inelegant and wasteful), so I instead tried the following 
> request:
> 
> http://192.168.1.1:8080/txtapi/show?calls
> 
> That works better, but then another wrinkle appears. Caller ID (and Callee 
> ID) names sometimes have commas, which messes up the CSV.
> 
> This appears to be a bug in mod_commands.c (or I suppose you could call it 
> an artifact of somebody coding up something quickly and not worrying about 
> the absolute correctness of something that is not critical to the core). 
> There is also another bug with show calls via webapi in that the generated 
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due 
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally, 
> another bug (or maybe just unimplemented feature) is that xmlapl isn't 
> really different from webapi.
> 
> These problems occur in the latest source tree (20100301).
> 
> I am wondering if (a) one of the developers can fix these or (b) if I fix 
> the bugs, how can I submit patches [and if I submit patches, will they be 
> accepted into the main tree]?
> 
> The fixes are pretty trivial and I'd be happy to code them up if somebody 
> will tell me how I can submit the patches (haven't done it before).

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From msc at freeswitch.org  Mon Mar  1 13:52:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 13:52:41 -0800
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>

On Mon, Mar 1, 2010 at 1:31 PM, Wellie Chao <wchao at yahoo.com> wrote:

> I am using the xml_rpc interface like so:
>
> http://192.168.1.1:8080/webapi/show?calls
>
> I get a response in a table, which is a little bit undesirable because I
> want to parse the result in a program and don't want to parse a bunch of
> HTML (just seems inelegant and wasteful), so I instead tried the following
> request:
>
> http://192.168.1.1:8080/txtapi/show?calls
>
> That works better, but then another wrinkle appears. Caller ID (and Callee
> ID) names sometimes have commas, which messes up the CSV.
>
> This appears to be a bug in mod_commands.c (or I suppose you could call it
> an artifact of somebody coding up something quickly and not worrying about
> the absolute correctness of something that is not critical to the core).
> There is also another bug with show calls via webapi in that the generated
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
> another bug (or maybe just unimplemented feature) is that xmlapl isn't
> really different from webapi.
>
> These problems occur in the latest source tree (20100301).
>
> I am wondering if (a) one of the developers can fix these or (b) if I fix
> the bugs, how can I submit patches [and if I submit patches, will they be
> accepted into the main tree]?
>
> The fixes are pretty trivial and I'd be happy to code them up if somebody
> will tell me how I can submit the patches (haven't done it before).
>

Go to jira.freeswitch.org and sign up for an account. You can open an issue,
assign it to yourself, and then submit a patch. Be sure to upload .txt files
and not tar, gz, or bz2! Also, check out the bug reporting tips on this
page:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

It gives some more details on reporting bugs and the things that you should
be doing.
-MC
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From brian at freeswitch.org  Mon Mar  1 13:56:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 15:56:32 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>
Message-ID: <6E086D9C-0968-4E86-B954-8247FD6B32E4@freeswitch.org>

He should be doing "show channels as xml" via XML API if he wants to parse them. .. or better yet doing it via real XMLRPC.  ;)

/b

On Mar 1, 2010, at 3:52 PM, Michael Collins wrote:

> Go to jira.freeswitch.org and sign up for an account. You can open an issue, assign it to yourself, and then submit a patch. Be sure to upload .txt files and not tar, gz, or bz2! Also, check out the bug reporting tips on this page:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> It gives some more details on reporting bugs and the things that you should be doing.
> -MC
> 

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From andrew at hijacked.us  Mon Mar  1 14:15:24 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 1 Mar 2010 17:15:24 -0500
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8C32D7.40202@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
Message-ID: <20100301221524.GB1751@hijacked.us>

On Mon, Mar 01, 2010 at 03:34:15PM -0600, Mark Sobkow wrote:
> I'm using the originate command from Erlang, specifying a timeout of 30 
> seconds.  It is properly giving up on dialing, but it's still returning 
> a call UUID instead of an error/timed out status.
> 
How are you doing an originate, and what 'UUID' are you getting back?

The way I do it is I do an bgapi originate with origination_uuid set and
I poll for that UUID 10 times every 100 milliseconds to see if its up.
Even if you originate a channel that's going to ring out the channel
will be up for that period and so you can connect to it and receive the
HANGUP event with the cause code to see what happened.

See the init() function in

http://github.com/Vagabond/OpenACD/blob/master/src/freeswitch_ring.erl

for an example. Its ugly but effective.

Andrew


From msc at freeswitch.org  Mon Mar  1 14:29:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 14:29:18 -0800
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
Message-ID: <87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>

On Sun, Feb 28, 2010 at 3:36 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Is there anyway to detect energy levels on a channel that is being
> controlled by mod_lua? Please point me in the right direction if there is.
> Thanks.
>
> Is there a way to do it on a channel that is not controlled by Lua?!
-MC
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From robert.hadley at teotech.com  Mon Mar  1 14:44:21 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 1 Mar 2010 14:44:21 -0800
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
	FXOline
In-Reply-To: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
References: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
Message-ID: <7524ABB6A6684383A76BB9E8B55C679A@greyhawk.tonecommander.com>

 

Using a Sangoma A200 FXO card, is there any way for Freeswitch to
automatically detect a disconnected CO line and use another FXO channel?

 

Thanks,

Robert

 

  _____  

From: Robert Hadley [mailto:robert.hadley at teotech.com] 
Sent: Thursday, February 25, 2010 4:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
FXOline

 

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
channel is disconnected and tries dialing out on the channel anyway.  No
error is reported.  The call doesn't timeout until a minute later.
Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next
connected channel?

 

I have configured a Sangoma A200 FXO card as a FXO span.

 

[span wanpipe FXO]

name => PSTN Line 1

number => 4253491059

fxo-channel => 2:3

name => PSTN Line 2

number => 4253491058

fxo-channel => 2:4

 

 

The wanpipe driver does detect and report when a CO line is connected or
disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
reported in the log.

 

/var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
FXO Line is disconnected!

FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_TRAP][3:1] STATE [DOWN]

 

/var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
FXO Line is connected!

FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_CLEAR][3:2] STATE [DOWN]

 

 

I have the dialplan configured to use the next available port in the FXO
span (there will be more than 2 channels later).

 

<extension name="OutgoingFXO">

  <condition field="destination_number" expression="^7(\d+)$">

    <action application="bridge" data="openzap/FXO/a/$1"/>

  </condition>

</extension>

 

 

Here is a portion of the log when that shows dialing out on a disconnected
analog FXO channel.

 

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/FXO/a/93491045)

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
channel OpenZAP/3:1/93491045

2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045)
State Change CS_NEW -> CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
from DOWN to DIALING

2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
OpenZAP/3:1/93491045 does not support the proxy feature, disabling.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
starting.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
handler on 3:1 for DIALING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
State Change CS_INIT -> CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
CHANNEL ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep

 

 

Thanks,

Robert

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From mrene_lists at avgs.ca  Mon Mar  1 14:59:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 17:59:31 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
Message-ID: <0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>

An easy way is to do an average of the absolute value of each L16  
sample over a period of time. Im not quite sure how to access audio  
channels from within Lua but if you are in C you can use  
switch_core_session_read_frame() to get audio frames.

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




On 1-Mar-10, at 5:29 PM, Michael Collins wrote:

>
>
> On Sun, Feb 28, 2010 at 3:36 AM, Matthew Fong <mattdfong at gmail.com>  
> wrote:
> Is there anyway to detect energy levels on a channel that is being  
> controlled by mod_lua? Please point me in the right direction if  
> there is. Thanks.
>
> Is there a way to do it on a channel that is not controlled by Lua?!
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Mar  1 15:24:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 17:24:16 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>

Perhaps you may be reading up on something quickly and not worrying about
the absolute correctness of something that is not critical to our time to
explain.

The proper syntax for the command is "show calls as xml" and you are not
really using XMLRPC you are hitting it via direct url. so you will then
need.

http://192.168.1.1:8080/txtapi/show?calls%20as%20xml<http://192.168.1.1:8080/txtapi/show?calls>


anything in the show command can be replied to with "as xml"

also, for fun, try

show calls as csv :::




On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:

> I am using the xml_rpc interface like so:
>
> http://192.168.1.1:8080/webapi/show?calls
>
> I get a response in a table, which is a little bit undesirable because I
> want to parse the result in a program and don't want to parse a bunch of
> HTML (just seems inelegant and wasteful), so I instead tried the following
> request:
>
> http://192.168.1.1:8080/txtapi/show?calls
>
> That works better, but then another wrinkle appears. Caller ID (and Callee
> ID) names sometimes have commas, which messes up the CSV.
>
> This appears to be a bug in mod_commands.c (or I suppose you could call it
> an artifact of somebody coding up something quickly and not worrying about
> the absolute correctness of something that is not critical to the core).
> There is also another bug with show calls via webapi in that the generated
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
> another bug (or maybe just unimplemented feature) is that xmlapl isn't
> really different from webapi.
>
> These problems occur in the latest source tree (20100301).
>
> I am wondering if (a) one of the developers can fix these or (b) if I fix
> the bugs, how can I submit patches [and if I submit patches, will they be
> accepted into the main tree]?
>
> The fixes are pretty trivial and I'd be happy to code them up if somebody
> will tell me how I can submit the patches (haven't done it before).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tculjaga at gmail.com  Mon Mar  1 15:37:42 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Mar 2010 00:37:42 +0100
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
Message-ID: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>

what about changing the destination ports on FS .. 5080 instead of 5060 ...
and make your phone register on that?

I think it could trick the router.

T.

On Mon, Mar 1, 2010 at 9:51 PM, Brian West <brian at freeswitch.org> wrote:

> The only disable button on a netgear is to a sledge hammer and go buy
> something that doesn't mess with your SIP traffic  :P
>
> /b
>
> On Mar 1, 2010, at 2:44 PM, Tom wrote:
>
> > MC,
> > the phone are both Aastra they are behind a netgear fvs338 which from
> > your post are very bad.  I will try to disable the ALG if i can and try
> > again.
> >
> > Thanks for the help
> >
> > Cheers
> > Tom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  1 15:52:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 17:52:05 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
Message-ID: <77FA7E8F-EACD-4262-A27D-ED11674E82B5@freeswitch.org>

Unless its EVIL and looks at all ports like some of them do.  :P

/b

On Mar 1, 2010, at 5:37 PM, Tihomir Culjaga wrote:

> what about changing the destination ports on FS .. 5080 instead of 5060 ... and make your phone register on that?
> 
> I think it could trick the router.
> 
> T.



From msc at freeswitch.org  Mon Mar  1 16:15:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 16:15:12 -0800
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
Message-ID: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>

On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> An easy way is to do an average of the absolute value of each L16 sample
> over a period of time. Im not quite sure how to access audio channels from
> within Lua but if you are in C you can use switch_core_session_read_frame()
> to get audio frames.
>

Show off! :) Related question: do media bugs allow you to access this kind
of information? Just curious...
-MC
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From mrene_lists at avgs.ca  Mon Mar  1 16:20:47 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 19:20:47 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
Message-ID: <8D25A218-B53B-48B2-BBF3-2E6ABA9E161B@avgs.ca>

Media bugs allow you to get a channel's audio so you can use that yeah.

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




On 1-Mar-10, at 7:15 PM, Michael Collins wrote:

>
>
> On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> An easy way is to do an average of the absolute value of each L16  
> sample over a period of time. Im not quite sure how to access audio  
> channels from within Lua but if you are in C you can use  
> switch_core_session_read_frame() to get audio frames.
>
> Show off! :) Related question: do media bugs allow you to access  
> this kind of information? Just curious...
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tomb at cachecomm.com  Mon Mar  1 16:24:14 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 17:24:14 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
Message-ID: <4B8C5AAE.2030908@cachecomm.com>

Tihomir Culjaga wrote:
> what about changing the destination ports on FS .. 5080 instead of 
> 5060 ... and make your phone register on that?
>
> I think it could trick the router.
>
> T.
>
> On Mon, Mar 1, 2010 at 9:51 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     The only disable button on a netgear is to a sledge hammer and go
>     buy something that doesn't mess with your SIP traffic  :P
>
>     /b
>
>     On Mar 1, 2010, at 2:44 PM, Tom wrote:
>
>     > MC,
>     > the phone are both Aastra they are behind a netgear fvs338 which
>     from
>     > your post are very bad.  I will try to disable the ALG if i can
>     and try
>     > again.
>     >
>     > Thanks for the help
>     >
>     > Cheers
>     > Tom
>
>
Tihomir,
I have tried 5080, 5090  and 5080 still the same result.


Tom 


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



From brian at freeswitch.org  Mon Mar  1 16:27:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 18:27:14 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8C5AAE.2030908@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
Message-ID: <45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>

Sledge hammer it is.

/b

On Mar 1, 2010, at 6:24 PM, Tom wrote:

> Tihomir,
> I have tried 5080, 5090  and 5080 still the same result.
> 
> 
> Tom 



From peder at networkoblivion.com  Mon Mar  1 16:31:14 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 1 Mar 2010 18:31:14 -0600
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel
	with	mod_lua
In-Reply-To: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
Message-ID: <070d01cab99f$aac3ae90$004b0bb0$@com>

What's a "media bug"?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 01, 2010 6:15 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Detecting Energy Levels on a Channel with
mod_lua

 

 

On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

An easy way is to do an average of the absolute value of each L16 sample
over a period of time. Im not quite sure how to access audio channels from
within Lua but if you are in C you can use switch_core_session_read_frame()
to get audio frames.


Show off! :) Related question: do media bugs allow you to access this kind
of information? Just curious...
-MC

 

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From mrene_lists at avgs.ca  Mon Mar  1 16:35:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 19:35:31 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <070d01cab99f$aac3ae90$004b0bb0$@com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
	<070d01cab99f$aac3ae90$004b0bb0$@com>
Message-ID: <D6E7975C-55FD-496A-A2A6-6221B13D11D3@avgs.ca>

A media bug is a way to get a channel's audio inside a module, you can  
have read, write and read_replace/write_replace bugs (those can  
actually change the audio).

Its what record_session uses to be able to record the call to a file  
while its executing other applications.
Other examples are... tone_detect, start_dtmf, eavesdrop...

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




On 1-Mar-10, at 7:31 PM, Peder wrote:

> What?s a ?media bug??
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Monday, March 01, 2010 6:15 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Detecting Energy Levels on a Channel  
> with mod_lua
>
>
>
> On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> An easy way is to do an average of the absolute value of each L16  
> sample over a period of time. Im not quite sure how to access audio  
> channels from within Lua but if you are in C you can use  
> switch_core_session_read_frame() to get audio frames.
>
> Show off! :) Related question: do media bugs allow you to access  
> this kind of information? Just curious...
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

Hello folks!

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

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

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

-Michael
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From david.yatsin at gmail.com  Mon Mar  1 17:34:12 2010
From: david.yatsin at gmail.com (Davud Yat Sin)
Date: Mon, 1 Mar 2010 20:34:12 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <4b8c6b16.9753f10a.4c0f.ffff9ce5@mx.google.com>

Hi lakshmanan, 
This was fixed in Openzap revision:1047

Can you update your openzap and update to these wanpipe drivers:
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.6.smg_pri-v1.63.tgz

 

 

David Yat Sin, BEng, Software Engineer

Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 119 | e. dyatsin at sangoma.com
www.sangoma.com  | wiki.sangoma.com

 

 

Dear all,
I'm having a A102 Sangoma hardware. I configured it with freeswitch.
wanrouter status, says both the port as connected.
My smg_prid version is
 
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
restart=============
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
1.54                           =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
2010                       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
15288                =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
===========================================
 
My freeswitch version is 16729.
I started freeswitch.
 
oz list
+OK
span: 1 (smg_prid)
type: Sangoma (boost)
chan_count: 60
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
 
I originated a call as
originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
 
But when I issued the following command:
originate openzap/smg_prid/a/9952248266
&bridge(openzap/smg_prid/a/8122133885)
It rings my mobile (9952248266) first, but after that the following error
was displayed
 
2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
The call got ended in my mobile.
 
Freeswitch log and smg_pri.conf
          http://pastebin.freeswitch.org/12248
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31
 
openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
  </boost_spans>
</configuration>
 
Please guide me to setup this one!!.

 

 

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From pjintheusa at gmail.com  Mon Mar  1 18:35:28 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Mar 2010 21:35:28 -0500
Subject: [Freeswitch-users] .NET / ESL
Message-ID: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>

Hi there,

I am sure that I have read somewhere that someone is developing a new .NET
library for FreeSWITCH ESL. If so can someone point me to some info / code
etc. so I can take a look.

Thanks

pj
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From brian at freeswitch.org  Mon Mar  1 18:42:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 20:42:41 -0600
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
References: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
Message-ID: <81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>

http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/

in the managed folder.

/b

On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:

> Hi there,
> 
> I am sure that I have read somewhere that someone is developing a new .NET library for FreeSWITCH ESL. If so can someone point me to some info / code etc. so I can take a look.
> 
> Thanks
> 
> pj
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Mon Mar  1 19:01:48 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Mar 2010 22:01:48 -0500
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>
References: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
	<81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>
Message-ID: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>

Thanks

On Mon, Mar 1, 2010 at 9:42 PM, Brian West <brian at freeswitch.org> wrote:

> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
>
> in the managed folder.
>
> /b
>
> On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
>
> > Hi there,
> >
> > I am sure that I have read somewhere that someone is developing a new
> .NET library for FreeSWITCH ESL. If so can someone point me to some info /
> code etc. so I can take a look.
> >
> > Thanks
> >
> > pj
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From delorenzodesign at gmail.com  Mon Mar  1 19:17:50 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Mon, 1 Mar 2010 22:17:50 -0500
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback doesn't seem to get called
Message-ID: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>

I've got the following Lua script working in a sense, but the InputCallback
never seems to get called while the file is being streamed to the call
recipient.  I've tried moving the "vmd start" command and set input callback
around a bit, but to no avail.  I'm testing this against a cell phone voice
mailbox (Verizon).

freeswitch.consoleLog("info","########################################################\n\n");
>
> number_to_call = argv[1]
> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>
> voicemail_detected = false;
>
> function onInput(s, type, obj)
>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
> *************\n");
>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
> session:getVariable("vmd_detect") .. " *************\n");
>
>     if(type == "event" and voicemail_detected == false) then
>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
> MACHINE DETECTED *************\n");
>         voicemail_detected = true;
>         return "break";
>     end
> end
>
> function playbackMessage()
>     sleep_time = 1000;
>     if(voicemail_detected) then
>        sleep_time = 2500;
>     end
>     -- sleep a second
>     session:sleep(sleep_time);
>     -- play a file
>     session:streamFile(message_to_play);
> end
>
> function notify()
>     session =
> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
> .. number_to_call)
>
>
>
>     if(session:ready()) then
>         -- answer the call
>         session:answer();
>         session:setInputCallback("onInput", "");
>         session:execute("vmd","start");
>
>         playbackMessage();
>         if(voicemail_detected) then
>             freeswitch.consoleLog("notice","************ DOING PLAYBACK FOR
> VOICEMAIL/ANSWERING MACHINE *************\n");
>             playbackMessage();
>         end
>
>         freeswitch.consoleLog("notice", "********* hanging up session
> **********\n");
>         -- hangup
>         session:hangup();
>     end
> end
>
> notify();
>
>
> freeswitch.consoleLog("info","########################################################\n\n");
>

-- 
Michael De Lorenzo
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From wchao at yahoo.com  Mon Mar  1 19:29:24 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Mon, 1 Mar 2010 22:29:24 -0500 (EST)
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>

Terrific. "show calls as xml" works great and just what I needed.

P.S. "show calls as csv" still has the comma bug. Not a big deal since I 
can get it as XML. CSV needs quotes around field values that contain a 
comma. Pretty easy to fix. Few lines of C (strchr for "," and if found, 
put quotes around field value).


Date: Mon, 1 Mar 2010 17:24:16 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bug in mod_commands.c with show calls command

Perhaps you may be reading up on something quickly and not worrying about
the absolute correctness of something that is not critical to our time to explain.

The proper syntax for the command is "show calls as xml" and you are not really using XMLRPC you are hitting it via direct url.
so you will then need.

http://192.168.1.1:8080/txtapi/show?calls%20as%20xml


anything in the show command can be replied to with "as xml"

also, for fun, try

show calls as csv :::




On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
      I am using the xml_rpc interface like so:

      http://192.168.1.1:8080/webapi/show?calls

      I get a response in a table, which is a little bit undesirable because I
      want to parse the result in a program and don't want to parse a bunch of
      HTML (just seems inelegant and wasteful), so I instead tried the following
      request:

      http://192.168.1.1:8080/txtapi/show?calls

      That works better, but then another wrinkle appears. Caller ID (and Callee
      ID) names sometimes have commas, which messes up the CSV.

      This appears to be a bug in mod_commands.c (or I suppose you could call it
      an artifact of somebody coding up something quickly and not worrying about
      the absolute correctness of something that is not critical to the core).
      There is also another bug with show calls via webapi in that the generated
      HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
      to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
      another bug (or maybe just unimplemented feature) is that xmlapl isn't
      really different from webapi.

      These problems occur in the latest source tree (20100301).

      I am wondering if (a) one of the developers can fix these or (b) if I fix
      the bugs, how can I submit patches [and if I submit patches, will they be
      accepted into the main tree]?

      The fixes are pretty trivial and I'd be happy to code them up if somebody
      will tell me how I can submit the patches (haven't done it before).

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




--
Anthony Minessale II

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

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

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



From infos at madovsky.org  Mon Mar  1 20:38:40 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 1 Mar 2010 23:38:40 -0500
Subject: [Freeswitch-users] server to server jingle
Message-ID: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>

Hi all,

is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
I read this article
http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
but no example
and this one
http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
but seems to doesn't work as expected.

Thanks !

Franck
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From msc at freeswitch.org  Mon Mar  1 21:07:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 21:07:22 -0800
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback 	doesn't seem to get called
In-Reply-To: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
Message-ID: <87f2f3b91003012107w59b21e6m2a86100866bb14d3@mail.gmail.com>

I haven't played with events inside Lua yet, but here's an example:
http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts

Look at the onInput() function in that example. There it gives the syntax
for extracting the event header to see what event type has come in.

The other question I have: did you confirm that mod_vmd triggers properly
when going to this mailbox? You could write a simple dialplan that sets
mod_vmd, calls the phone, sleeps for 10 or so seconds, and then dumps the
var ${vmd_detect} to see what it says. I've used mod_vmd from the XML
dialplan with a fair amount of success, although to my knowledge I never
actually tried it with a Verizon wireless VM box.

Confirm that mod_vmd works with just the dialplan before trying to debug the
Lua script.

-MC

On Mon, Mar 1, 2010 at 7:17 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I've got the following Lua script working in a sense, but the InputCallback
> never seems to get called while the file is being streamed to the call
> recipient.  I've tried moving the "vmd start" command and set input callback
> around a bit, but to no avail.  I'm testing this against a cell phone voice
> mailbox (Verizon).
>
>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>>  number_to_call = argv[1]
>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>
>> voicemail_detected = false;
>>
>> function onInput(s, type, obj)
>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>> *************\n");
>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>> session:getVariable("vmd_detect") .. " *************\n");
>>
>>      if(type == "event" and voicemail_detected == false) then
>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>> MACHINE DETECTED *************\n");
>>         voicemail_detected = true;
>>         return "break";
>>     end
>> end
>>
>> function playbackMessage()
>>     sleep_time = 1000;
>>     if(voicemail_detected) then
>>        sleep_time = 2500;
>>     end
>>     -- sleep a second
>>     session:sleep(sleep_time);
>>     -- play a file
>>     session:streamFile(message_to_play);
>> end
>>
>> function notify()
>>     session =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>> .. number_to_call)
>>
>>
>>
>>     if(session:ready()) then
>>         -- answer the call
>>         session:answer();
>>         session:setInputCallback("onInput", "");
>>         session:execute("vmd","start");
>>
>>         playbackMessage();
>>         if(voicemail_detected) then
>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>             playbackMessage();
>>         end
>>
>>         freeswitch.consoleLog("notice", "********* hanging up session
>> **********\n");
>>         -- hangup
>>         session:hangup();
>>     end
>> end
>>
>> notify();
>>
>>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>
> --
> Michael De Lorenzo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gkuri at ieee.org  Mon Mar  1 21:49:06 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 1 Mar 2010 21:49:06 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
	<191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
Message-ID: <8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>

I've torn my hair out for hours and I still can't get it to work with
the domain name set everywhere <sigh> ...

I can get it to work if I hardcode the IP address on the phones and in
all the config files, but I can't get the domain name to work on
incoming calls, I'm at a total loss. If I had a mismatch somewhere,
I'd think it wouldn't work on outgoing calls either, but that seems to
work fine.

Any other ideas? Might it have anything to do with the incoming call
coming in from the PSTN and going through the external profile?

Cheers,
Gabe

On Mon, Mar 1, 2010 at 1:22 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I really want to be done with this thread......
>
> You were using the IP in some places and the hostnames in others, this makes
> it so the SQL statements not match because sql statements cannot do dns
> lookups.? I will not be doing any work to resolve hostnames back and forth
> to IP inside FS.
>
>
> The subscribes were using the hostname and the db and invites were using the
> IP.
>
> you have to have the same exact domain in your user directory and have the
> force-register-domain and force-register-domain-db also set to that value or
> omit them completely and use the domain names in 100% of the config fields
> on the phone.
>
>
>
>
> On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Hi Anthony,
>>
>> I appreciate all the hard work yourself, Brian, Mike and all the other
>> developers have put in FS! By no means am I discounting any of it,
>> it's truly invaluable.
>>
>> Our "philosophy" has always been to use DNS where available, so that
>> we're not worrying about re-configuring a bunch of devices if IP
>> addresses change and this obviously goes to setting the phones up as
>> well to use DNS for the proxy and registrar fields. And it allows us
>> to do fancy things, like split-view DNS, where if a phone is taken
>> off-site and we want it to remotely register to FS, it will get the
>> public IP address of the FS server, rather than the private, without
>> re-configuring the phone.
>>
>> If I'm understanding your email correctly, you're saying we need to
>> use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
>> DNS, but perhaps I'm missing something somewhere. How would we be able
>> to use the DNS name of our SIP domain in phones (rather than IP) and
>> ensure SLA works for both incoming/outgoing calls? Is that possible at
>> this point in time or do we need to use the IP addresses in the proxy
>> field on the phone if we want to use SLA?
>>
>> Thanks for all your help.
>>
>> Gabriel Kuri
>>
>>
>> On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > you can use names if you use *all names*
>> > you just can't mix, but all IP is much faster and safer because there is
>> > no
>> > need to do constant dns lookups on something that never changes anyway,
>> > you
>> > can easily alias your profile to a friendly name, it also makes it more
>> > portable.
>> >
>> >
>> >
>> >
>> > On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine
>> > <yehavi.bourvine at gmail.com>
>> > wrote:
>> >>
>> >> Hello Anthony,
>> >>
>> >> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>> >>>
>> >>> I am trying my best not to be completely annoyed at the "somewhat
>> >>> limiting" remark after going over the countless hours I spent making
>> >>> sure
>> >>> this feature work and then being patient with 2 people insisting there
>> >>> was a
>> >>> bug when they both shared an identical misconfiguration.
>> >>
>> >> "somewhat limiting" was because my experiments up to now with Polycom
>> >> and
>> >> SIP/TLS were successfull with domain names only. However, I am going to
>> >> inverstigate this further to see how this can work with IP only.
>> >>
>> >> In the meantime I've documented the required configuration under the
>> >> Polycom wiki page.
>> >>
>> >>>
>> >>> If we started doing DNS lookups everywhere so hostname could be
>> >>> interchanged with IP we would have large blocking all over the place
>> >>> when
>> >>> the DNS was not available and horribly messy code doing all the
>> >>> lookups.
>> >>> There was already an experiment called project asterisk that proved
>> >>> the
>> >>> flaws in this design approach.
>> >>>
>> >> I think that's why some of us move from Asterisk to FreeSwitch. I also
>> >> preffer less to no DNS lookups on a critical system which must?work
>> >> even
>> >> when there are some kinds of communication problems.
>> >>
>> >> ????????????????????? Thanks, __Yehavi:
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Mon Mar  1 22:01:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 00:01:38 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
	<191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
	<8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>
Message-ID: <3DE15C61-E4E3-4EC2-A319-20350092C8F0@freeswitch.org>

You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen.

Verify the subscribe packet...

/b

On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:

> I've torn my hair out for hours and I still can't get it to work with
> the domain name set everywhere <sigh> ...
> 
> I can get it to work if I hardcode the IP address on the phones and in
> all the config files, but I can't get the domain name to work on
> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
> I'd think it wouldn't work on outgoing calls either, but that seems to
> work fine.
> 
> Any other ideas? Might it have anything to do with the incoming call
> coming in from the PSTN and going through the external profile?
> 
> Cheers,
> Gabe



From infos at madovsky.org  Mon Mar  1 22:05:17 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 01:05:17 -0500
Subject: [Freeswitch-users] wiki broken link pocketsphinx
Message-ID: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>

Hello,

the link that goes to grammar sphinx is broken.
the right link is

http://cmusphinx.sourceforge.net/sphinx4/

Regards

Franck
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From infos at madovsky.org  Mon Mar  1 22:24:35 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 01:24:35 -0500
Subject: [Freeswitch-users] last svn version compile error
Message-ID: <5887BE899D4F47B0BF05FCB3340B58CB@MOBILEE1705>

Hi,

Ino succes to update freeswitch to version 16859
with usual command "make current" etc...
but now I got this error :

/usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of ?switch_core_media_bug_add? makes integer from pointer without a cast
/usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function ?switch_core_media_bug_add?

Any idea ?

thanks

Franck

Linux Fedora 10  64bits
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From dujinfang at gmail.com  Mon Mar  1 22:57:51 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Mar 2010 14:57:51 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
Message-ID: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>

I have iLBC configured in profile internal

CODECS IN               PCMU,PCMA,iLBC at 30i,GSM
CODECS OUT              PCMU,PCMA,iLBC at 30i,GSM

and     <param name="inbound-codec-negotiation" value="generous"/>

but it seems not comparing with iLBC

 010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[GSM:3:8000:20]


can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .

Thanks.


From dujinfang at gmail.com  Mon Mar  1 23:01:28 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Mar 2010 15:01:28 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
In-Reply-To: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
Message-ID: <23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>

sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)


2010/3/2 Seven Du <dujinfang at gmail.com>:
> I have iLBC configured in profile internal
>
> CODECS IN ? ? ? ? ? ? ? PCMU,PCMA,iLBC at 30i,GSM
> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,iLBC at 30i,GSM
>
> and ? ? <param name="inbound-codec-negotiation" value="generous"/>
>
> but it seems not comparing with iLBC
>
> ?010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
> payload to 101
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>
>
> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>
> Thanks.
>


From lakindia89 at gmail.com  Mon Mar  1 23:14:28 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 2 Mar 2010 12:44:28 +0530
Subject: [Freeswitch-users] make current - System Hangs
Message-ID: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>

Hi all,
I did make current from my freeswitch src directory, but the system got
hanged.
Any one know why this happened.

Here is the output that is printed when I run make current
http://pastebin.freeswitch.org/12279
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From tculjaga at gmail.com  Tue Mar  2 01:03:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Mar 2010 10:03:27 +0100
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
Message-ID: <65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>

bastard !!!! :)

T.

On Tue, Mar 2, 2010 at 1:27 AM, Brian West <brian at freeswitch.org> wrote:

> Sledge hammer it is.
>
> /b
>
> On Mar 1, 2010, at 6:24 PM, Tom wrote:
>
> > Tihomir,
> > I have tried 5080, 5090  and 5080 still the same result.
> >
> >
> > Tom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sharad at coraltele.com  Tue Mar  2 03:20:31 2010
From: sharad at coraltele.com (Sharad)
Date: Tue, 2 Mar 2010 03:20:31 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch & T.38
Message-ID: <1267528831989-4660389.post@n2.nabble.com>


Hi friends,

Is Freeswitch 1.05 carrying T.38 support also ? 

If not, do we have any community member who can provide paid T.38 support
with Freeswitch ?


-- 
View this message in context: http://n2.nabble.com/Freeswitch-T-38-tp4660389p4660389.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From michal.kalinowski at interia.pl  Tue Mar  2 03:28:47 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Tue, 2 Mar 2010 12:28:47 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
	<7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
	<87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>
Message-ID: <7c74f5761003020328u3a480d59v703899d3fc81a1fb@mail.gmail.com>

Thanks Michael right now it's working correct.
In wiki example there is a lack of args "tts_engine" "tts_voice" and
"max_timeouts"

BR,
Micha?

2010/3/1 Michael Collins <msc at freeswitch.org>:
>
>
> 2010/3/1 michal kalinowski <michal.kalinowski at interia.pl>
>>
>> Thank's Brian
>>
>> So at this moment i have only clean configuration in lua without any XML.
>>
>> I get example from wiki:
>>
>> hash={["main"]=undef,
>>
>> ? ["name"]="top",
>>
>> ? ? ? ? ["greet_long"]="phrase:demo_ivr_main_menu",
>>
>> ["greet_short"]="phrase:demo_ivr_main_menu_short",
>>
>> ["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",
>>
>> ["exit_sound"]="voicemail/vmgoodbye.wav",
>>
>> ["confirm_macro"]="undef",
>>
>> ?["confirm_key"]="undef",
>>
>> ? ?["confirm_attempts"]="3",
>>
>> ? ? ?["inter_digit_timeout"]="2000",
>>
>> ? ? ? ["digit_len"]="1",
>>
>> ? ? ? ? ? ? ? ["timeout"]="10000",
>> ["max_failures"]="3"
>> }
>> top = freeswitch.IVRMenu(hash["main"],
>> hash["name"],
>> hash["greet_long"],hash["greet_short"],
>> hash["invalid_sound"],
>> hash["exit_sound"],
>> hash["confirm_macro"],
>> hash["confirm_key"],
>> hash["confirm_attempts"],
>> hash["inter_digit_timeout"],
>> hash["digit_len"],
>> hash["timeout"],
>> hash["max_failures"]);
>>
>> top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
>> top:execute(session, "top");
>>
>> But something is wrong freeswitch say
>> 2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
>> expected 16..16 args, got 13
>> stack traceback:
>> ? ? ? ?[C]: in function 'IVRMenu'
>> ? ? ? ?/usr/local/freeswitch/scripts/ivr.lua:19: in main chunk
>>
>> What args FS expect?
>
> I ran into this issue when I was converting the demo IVR to Lua. There are
> indeed 16 arguments but you've only got 13 of them. Here's a snippet from
> the Lua script I'm working on:
> ivr_def = {
> ??? ["main"]??????????????? = undef,
> ??? ["name"]??????????????? = "demo_ivr_lua",
> ??? ["greet_long"]????????? = "phrase:demo_ivr_main_menu",
> ??? ["greet_short"]???????? = "phrase:demo_ivr_main_menu_short",
> ??? ["invalid_sound"]?????? = "ivr/ivr-that_was_an_invalid_entry.wav",
> ??? ["exit_sound"]????????? = "voicemail/vm-goodbye.wav",
> ??? ["confirm_macro"]?????? = "",
> ??? ["confirm_key"]???????? = "",
> ??? ["tts_engine"]????????? = "flite",
> ??? ["tts_voice"]?????????? = "rms",
> ??? ["confirm_attempts"]??? = "3",
> ??? ["inter_digit_timeout"] = "2000",
> ??? ["digit_len"]?????????? = "4",
> ??? ["timeout"]???????????? = "10000",
> ??? ["max_failures"]??????? = "3",
> ??? ["max_timeouts"]??????? = "2"
> }
> Those are all 16 arguments. Notice you need tts_engine, tts_voice, and
> max_timeouts. Try that and let us know how it goes.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yehavi.bourvine at gmail.com  Tue Mar  2 04:56:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 2 Mar 2010 14:56:53 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
	<191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
Message-ID: <d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>

Since I am aware that I am the only one who experience this problem, I am
trying to ivestigate it, although I do not know the source code well...

I removed all the datbase calls from our scripts and the problem still
persists. It happens usually when two calls arrive to the same extension
(usually to an SLA one) and one of them disconnects during the ringing
phase.

I've inserted some "printfs" in switch_core_sqldb.c and found that when this
happens the variable dbh->io_mutex has a null value, thus no mutex is locked
(as far as I understand it).

I hope this might give some clues.

                      Thanks! __Yehavi:

 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>

>  You probably had it right in the last email.
> Polling the core's db from lua is not recommended.
> We use sla extensively and never once see your issue.
>
>  On Feb 28, 2010 8:55 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
>  After a few more tests I *think* it is related to SLA. Since it cannot be
> reproduced consistenty I say "think".
>
> What I have is:
> - 80635 which is an SLA extension between two Polycoms.
> - 80636 which is a private extension on Polycom
> - 80632 which is a Cisco extension.
> - 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
> gateway doesn't hav any login information it is not defined as a gateway but
> accepted via ACL.
>
> The tests that passed ok:
> - From/to 80636/80632 to 80635 (i.e. - all internal).
> - From/to 80635 to 80636/80632 (i.e. - all internal)
> - From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
> extensions).
>
> The one that fails after a few attempts:
> - From 86111 to 80635 (i.e. from the gateway to a shared extension).
>
> I hope that this gives some more clues.
>
>                           Thanks, __Yehavi:
>
>  2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>
>
>
> >
> > Hello,
> >
> >   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
> (16841M). The...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dftoro at yahoo.com  Tue Mar  2 05:24:59 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 2 Mar 2010 05:24:59 -0800 (PST)
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
Message-ID: <251258.58331.qm@web33508.mail.mud.yahoo.com>

Hi Phillip Jones,

Any concerns, You tell me, I help you

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 3/1/10, Phillip Jones <pjintheusa at gmail.com> wrote:

> From: Phillip Jones <pjintheusa at gmail.com>
> Subject: Re: [Freeswitch-users] .NET / ESL
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, March 1, 2010, 10:01 PM
> Thanks
> 
> On Mon, Mar 1, 2010 at 9:42 PM,
> Brian West <brian at freeswitch.org>
> wrote:
> 
> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
> 
> 
> 
> in the managed folder.
> 
> 
> 
> /b
> 
> 
> 
> On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
> 
> 
> 
> > Hi there,
> 
> >
> 
> > I am sure that I have read somewhere that someone is
> developing a new .NET library for FreeSWITCH ESL. If so can
> someone point me to some info / code etc. so I can take a
> look.
> 
> >
> 
> > Thanks
> 
> >
> 
> > pj
> 
> >
> _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From infos at madovsky.org  Tue Mar  2 06:25:54 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 09:25:54 -0500
Subject: [Freeswitch-users] Freeswitch & T.38
References: <1267528831989-4660389.post@n2.nabble.com>
Message-ID: <4BB20F0B3FCB43DCAD902617DDA79EEC@MOBILEE1705>

google wiki media proxy freeswitch

----- Original Message ----- 
From: "Sharad" <sharad at coraltele.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 02, 2010 6:20 AM
Subject: [Freeswitch-users] Freeswitch & T.38


>
> Hi friends,
>
> Is Freeswitch 1.05 carrying T.38 support also ?
>
> If not, do we have any community member who can provide paid T.38 support
> with Freeswitch ?
>
>
> -- 
> View this message in context: 
> http://n2.nabble.com/Freeswitch-T-38-tp4660389p4660389.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From brian at freeswitch.org  Tue Mar  2 06:28:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:28:45 -0600
Subject: [Freeswitch-users] wiki broken link pocketsphinx
In-Reply-To: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
Message-ID: <5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>

No thats not right either.  Care to elaborate?

/b

On Mar 2, 2010, at 12:05 AM, Madovsky wrote:

> Hello,
>  
> the link that goes to grammar sphinx is broken.
> the right link is
>  
> http://cmusphinx.sourceforge.net/sphinx4/
>  
> Regards
>  
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar  2 06:32:05 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:32:05 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
	<191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
	<d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>
Message-ID: <0ADA063F-9E3E-4540-BB3B-48373211BD6A@freeswitch.org>

Yes accessing the db in the method you're doing is unsupport and not recommended.

/b

On Mar 2, 2010, at 6:56 AM, Yehavi Bourvine wrote:

> Since I am aware that I am the only one who experience this problem, I am trying to ivestigate it, although I do not know the source code well...
>  
> I removed all the datbase calls from our scripts and the problem still persists. It happens usually when two calls arrive to the same extension (usually to an SLA one) and one of them disconnects during the ringing phase.
>  
> I've inserted some "printfs" in switch_core_sqldb.c and found that when this happens the variable dbh->io_mutex has a null value, thus no mutex is locked (as far as I understand it).
>  
> I hope this might give some clues.
>  
>                       Thanks! __Yehavi:



From brian at freeswitch.org  Tue Mar  2 06:33:59 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:33:59 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
In-Reply-To: <23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
Message-ID: <074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>

Yes that is because its @30i, All codecs offered MUST be 30ms if you want to offer iLBC at 30i.  So add @30i to all but GSM since you can't do GSM @ 30.

/b
On Mar 2, 2010, at 1:01 AM, Seven Du wrote:

> sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)
> 
> 
> 2010/3/2 Seven Du <dujinfang at gmail.com>:
>> I have iLBC configured in profile internal
>> 
>> CODECS IN               PCMU,PCMA,iLBC at 30i,GSM
>> CODECS OUT              PCMU,PCMA,iLBC at 30i,GSM
>> 
>> and     <param name="inbound-codec-negotiation" value="generous"/>
>> 
>> but it seems not comparing with iLBC
>> 
>>  010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
>> payload to 101
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>> 
>> 
>> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>> 
>> Thanks.
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Mar  2 06:34:19 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:34:19 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
Message-ID: <FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>

I have called netgear worse.  :)

/b

On Mar 2, 2010, at 3:03 AM, Tihomir Culjaga wrote:

> bastard !!!! :)
> 
> T.



From jonas.gauffin at gmail.com  Tue Mar  2 07:02:47 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 16:02:47 +0100
Subject: [Freeswitch-users] Incorrect nonce
Message-ID: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>

Hello,

I got a problem with rejected calls due to 403 (happens sometimes).
My sip provider found the problem, the nonce used by FS is sometimes
incorrect (I got the trace from my sip provider).

A sip trace can be found here: http://pastebin.freeswitch.org/12280


2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
(sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_INIT
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
(sofia/external/0707728XXX) State INIT
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX
SOFIA INIT
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
(sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
(sofia/external/0707728XXX) State INIT going to sleep
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
(sofia/external/0707728XXX) State ROUTING
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX
SOFIA ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
(sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
(sofia/external/0707728XXX) State ROUTING going to sleep
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
(sofia/external/0707728XXX) State CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
(sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][0]
2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][0]
2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][100]
2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][100]
2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [terminated][407]
2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]

Regards,
  Jonas
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From m.sobkow at marketelsystems.com  Tue Mar  2 07:06:15 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 09:06:15 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <20100301221524.GB1751@hijacked.us>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
Message-ID: <4B8D2967.7010403@marketelsystems.com>

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From infos at madovsky.org  Tue Mar  2 07:09:38 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 10:09:38 -0500
Subject: [Freeswitch-users] wiki broken link pocketsphinx
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
	<5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>
Message-ID: <30970655E0CD476A95CF8382A2742084@MOBILEE1705>

apparently Diego updated it.
I don't know if I'm skilled enough to elaborate

  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 9:28 AM
  Subject: Re: [Freeswitch-users] wiki broken link pocketsphinx


  No thats not right either.  Care to elaborate?


  /b


  On Mar 2, 2010, at 12:05 AM, Madovsky wrote:


    Hello,

    the link that goes to grammar sphinx is broken.
    the right link is

    http://cmusphinx.sourceforge.net/sphinx4/

    Regards

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





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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tomb at cachecomm.com  Tue Mar  2 07:15:50 2010
From: tomb at cachecomm.com (Tom)
Date: Tue, 02 Mar 2010 08:15:50 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>	<4B8C5AAE.2030908@cachecomm.com>	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
	<FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
Message-ID: <4B8D2BA6.1030907@cachecomm.com>

Brian West wrote:
> I have called netgear worse.  :)
>
> /b
>
> On Mar 2, 2010, at 3:03 AM, Tihomir Culjaga wrote:
>
>   
>> bastard !!!! :)
>>
>> T.
>>     
>
>
>   
I removed the Netgear P.O.S . and put the phones behind a linux firewall 
all is good.
Thanks for the help

Tom

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



From m.sobkow at marketelsystems.com  Tue Mar  2 07:39:20 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 09:39:20 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8D2967.7010403@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
	<4B8D2967.7010403@marketelsystems.com>
Message-ID: <4B8D3128.6020807@marketelsystems.com>

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From moises.silva at gmail.com  Tue Mar  2 07:45:22 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Mar 2010 10:45:22 -0500
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
	FXO line
In-Reply-To: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
References: <Acq2eHpz1O0wasTJQ1GGlQN3irPlFA==>
	<4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1003020745h20a1fdccmc7486c6454b810d5@mail.gmail.com>

Please try latest openzap (svn update in libs/openzap). I commited some code
that may help you.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Thu, Feb 25, 2010 at 7:13 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
> channel is disconnected and tries dialing out on the channel anyway.  No
> error is reported.  The call doesn?t timeout until a minute later.
> Shouldn?t Freeswitch/Openzap skip over a disconnected channel to the next
> connected channel?
>
>
>
> I have configured a Sangoma A200 FXO card as a FXO span.
>
>
>
> [span wanpipe FXO]
>
> name => PSTN Line 1
>
> number => 4253491059
>
> fxo-channel => 2:3
>
> name => PSTN Line 2
>
> number => 4253491058
>
> fxo-channel => 2:4
>
>
>
>
>
> The wanpipe driver does detect and report when a CO line is connected or
> disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
> reported in the log.
>
>
>
> /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
> FXO Line is disconnected!
>
> FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_TRAP][3:1] STATE [DOWN]
>
>
>
> /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
> FXO Line is connected!
>
> FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_CLEAR][3:2] STATE [DOWN]
>
>
>
>
>
> I have the dialplan configured to use the next available port in the FXO
> span (there will be more than 2 channels later).
>
>
>
> <extension name="OutgoingFXO">
>
>   <condition field="destination_number" expression="^7(\d+)$">
>
>     <action application="bridge" data="openzap/FXO/a/$1"/>
>
>   </condition>
>
> </extension>
>
>
>
>
>
> Here is a portion of the log when that shows dialing out on a disconnected
> analog FXO channel.
>
>
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/FXO/a/93491045)
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
> channel OpenZAP/3:1/93491045
>
> 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
> OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269
> (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
> from DOWN to DIALING
>
> 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
> OpenZAP/3:1/93491045 does not support the proxy feature, disabling.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
> starting.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 3:1 for DIALING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
> State Change CS_INIT -> CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
> CHANNEL ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
> (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From scottferri09 at gmail.com  Tue Mar  2 07:53:09 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 2 Mar 2010 21:23:09 +0530
Subject: [Freeswitch-users] Inbound settings
Message-ID: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>

Hi,


     1) How to configure inbound settings in FS?
     2) If so, Please list down the steps that I require to do?
     3) If have more than 1 DID number (let;'s say 4 DID's
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From brian at freeswitch.org  Tue Mar  2 08:00:22 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 10:00:22 -0600
Subject: [Freeswitch-users] Inbound settings
In-Reply-To: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
References: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
Message-ID: <D9251A3E-3341-4928-84DB-FE242DE6CAAC@freeswitch.org>

Have you read over the wiki?  http://wiki.freeswitch.org/wiki/Getting_Started_Guide

/b

On Mar 2, 2010, at 9:53 AM, Scott Fernandez wrote:

> Hi,
>  
>  
>      1) How to configure inbound settings in FS?
>      2) If so, Please list down the steps that I require to do?
>      3) If have more than 1 DID number (let;'s say 4 DID's
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Tue Mar  2 08:02:12 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:02:12 -0500
Subject: [Freeswitch-users] Inbound settings
References: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
Message-ID: <860AD4D8E1F34919A28DF973FF2C7969@MOBILEE1705>

http://wiki.freeswitch.org/wiki/Installation_Guide

  ----- Original Message ----- 
  From: Scott Fernandez 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 10:53 AM
  Subject: [Freeswitch-users] Inbound settings


  Hi,


       1) How to configure inbound settings in FS?
       2) If so, Please list down the steps that I require to do?
       3) If have more than 1 DID number (let;'s say 4 DID's




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From m.sobkow at marketelsystems.com  Tue Mar  2 08:10:43 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 10:10:43 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8D3128.6020807@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
	<4B8D2967.7010403@marketelsystems.com>
	<4B8D3128.6020807@marketelsystems.com>
Message-ID: <4B8D3883.9040902@marketelsystems.com>

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From sos at sokhapkin.dyndns.org  Tue Mar  2 08:13:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 11:13:06 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
Message-ID: <201003021113.06820.sos@sokhapkin.dyndns.org>

FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No single 
SIP request hits it. What does it doing? I unloaded all modules which can be 
unloaded, no difference. Tried with both -vm and -heavy-timer start up 
options. I'm running gentoo linux.

# uname -a
Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686 Intel(R) 
Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux


From jonas.gauffin at gmail.com  Tue Mar  2 08:20:26 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 17:20:26 +0100
Subject: [Freeswitch-users] Diversion header
Message-ID: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>

Hello,

I got a incoming call from my provider which goes to a javascript. The
javascript transfers the call to an external destination (goes back through
my provider).
A diversion headers is added by FreeSWITCH on the outbound call. How can I
remove it (or if not possible, how do I change it)?

javascript:
        session.setCallerData("caller_id_number", aNumber);
session.setVariable("gate_caller_id_number", aNumber);
        session.setVariable("gate_site_id", SITE_ID);
        session.execute("transfer", this.dtmfBuffer + " XML internal");



send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200:
   ------------------------------------------------------------------------
   INVITE sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>SIP/2.0
   Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS
   Max-Forwards: 67
   From: "0236661XXX" <sip:0243795362 at 212.247.XX.XX>;tag=BHvytKtv47r2a
   To: <sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>
>
   Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5
   CSeq: 127654862 INVITE
   Contact: <sip:gw+dalaconnecttele2 at 212.247.XX.XX
:5060;transport=udp;gw=dalaconnecttele2>
   User-Agent: Gateon
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 273
   Diversion: 0 <sip:0 at 212.151.Y.Y
;user=phone>;reason=unconditional;counter=1;privacy=off
   X-FS-Support: update_display
   Remote-Party-ID: "0236661XXX" <sip:0243795362 at 212.247.XX.XX
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX
   s=FreeSWITCH
   c=IN IP4 212.247.XX.XX
   t=0 0
   m=audio 32712 RTP/AVP 8 3 101 13
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
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From infos at madovsky.org  Tue Mar  2 08:20:54 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:20:54 -0500
Subject: [Freeswitch-users] last svn version compile error
Message-ID: <C8664F4269084354BB7AB9203887E30E@MOBILEE1705>

The compilation is fine if I comment out mod_vmd from modules.conf

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 1:24 AM
  Subject: last svn version compile error


  Hi,

  Ino succes to update freeswitch to version 16859
  with usual command "make current" etc...
  but now I got this error :

  /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of ?switch_core_media_bug_add? makes integer from pointer without a cast
  /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function ?switch_core_media_bug_add?

  Any idea ?

  thanks

  Franck

  Linux Fedora 10  64bits
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From moises.silva at gmail.com  Tue Mar  2 08:35:47 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Mar 2010 11:35:47 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>

ps -LlFm -p `pidof freeswitch`

That shows you CPU usage per FreeSWITCH thread

Then

pstack `pidof freeswitch`

That shows the stack trace of every FreeSWITCH thread and then you can find
exactly what is doing.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Tue, Mar 2, 2010 at 11:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> single
> SIP request hits it. What does it doing? I unloaded all modules which can
> be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  2 08:42:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 10:42:56 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>

If you don't like that CPU usage, use a different kernel or disable the
broken "tickles" features.

you can't use both -vm and -heavy-timer because -heavy-timer replaces -vm
permanently.


On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> single
> SIP request hits it. What does it doing? I unloaded all modules which can
> be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  2 08:43:45 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:43:45 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <6FB1E9CF2C044BBCBE87D44335CD2383@MOBILEE1705>

Hi,

I searched some articles of FS in a cluster environment
but there are some unclear writes :

in http://www.technoclicks.com/article-2545.php from 2006 it says

"Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

and this thread from 2008
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
that describes contradiction of how to manage clustering.

so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

Regards

F
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From anthony.minessale at gmail.com  Tue Mar  2 08:44:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 10:44:23 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
Message-ID: <191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>

Are you reporting what you think is a bug on the mailing list?
I think you have been around long enough to know better.............


On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes
> incorrect (I got the trace from my sip provider).
>
> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>
>
> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_INIT
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/0707728XXX) State INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX
> SOFIA INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/0707728XXX) State INIT going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/0707728XXX) State ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
> sofia/external/0707728XXX SOFIA ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/0707728XXX) State ROUTING going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/0707728XXX) State CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [terminated][407]
> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>
> Regards,
>   Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar  2 08:53:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 10:53:35 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
Message-ID: <151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>

Jonas,
	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.

Remember the list is not a bug tracker... Thats what we have jira for  ;)

Thanks,
Brian

On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:

> Are you reporting what you think is a bug on the mailing list?
> I think you have been around long enough to know better.............
> 
> 
> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> Hello,
> 
> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
> 
> A sip trace can be found here: http://pastebin.freeswitch.org/12280
> 
> 
> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
> 
> Regards,
>   Jonas
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Tue Mar  2 08:56:52 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:56:52 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <76B2210A926D4D06B34B38BBE633210D@MOBILEE1705>

what FS team thinks about this SIP clustering architecture ?

http://blogs.sun.com/kshitiz/entry/converged_load_balancer

Regards

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 11:43 AM
  Subject: clustering question


  Hi,

  I searched some articles of FS in a cluster environment
  but there are some unclear writes :

  in http://www.technoclicks.com/article-2545.php from 2006 it says

  "Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

  and this thread from 2008
  http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
  that describes contradiction of how to manage clustering.

  so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

  Regards

  F
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From sos at sokhapkin.dyndns.org  Tue Mar  2 09:00:15 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 12:00:15 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>
Message-ID: <201003021200.16252.sos@sokhapkin.dyndns.org>

Tickless feature is disabled in the kernel.

$ zcat /proc/config.gz | grep CONFIG_NO_HZ
# CONFIG_NO_HZ is not set


On Tuesday 02 March 2010, Anthony Minessale wrote:
> If you don't like that CPU usage, use a different kernel or disable the
> broken "tickles" features.
>
> you can't use both -vm and -heavy-timer because -heavy-timer replaces -vm
> permanently.
>
>
> On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> > single
> > SIP request hits it. What does it doing? I unloaded all modules which can
> > be
> > unloaded, no difference. Tried with both -vm and -heavy-timer start up
> > options. I'm running gentoo linux.
> >
> > # uname -a
> > Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> > Intel(R)
> > Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From jonas.gauffin at gmail.com  Tue Mar  2 09:04:26 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 18:04:26 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
Message-ID: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>

You are both correct.

Issue created.

On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:

> Jonas,
> Also I need a complete pcap of this without modifications.  I can't hand
> calculate this to see what is going on without intact data. Please open a
> jira and email me the pcap in private if you must.
>
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>
> Thanks,
> Brian
>
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>
> Are you reporting what you think is a bug on the mailing list?
> I think you have been around long enough to know better.............
>
>
> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello,
>>
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes
>> incorrect (I got the trace from my sip provider).
>>
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>
>>
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>> (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>> sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>> (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>> (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>> sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>> (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>
>> Regards,
>>   Jonas
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar  2 09:06:20 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 12:06:20 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <C3B965932B59454EAC61E6EC90766510@MOBILEE1705>

also from 2007
http://ftp.iptel.org/pub/sems/fostel_2007_coeffic_sems_load_balancing.pdf
that says SEMS can make round robin and hash load balancing.
I don't talk here about failover since this can be resolved with conventional clustering solution
as hearbeat pacemaker redhat cluster etc... but especially load balancing to increase nujmber
of calls

Regards

F


  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 11:56 AM
  Subject: Re: clustering question


  what FS team thinks about this SIP clustering architecture ?

  http://blogs.sun.com/kshitiz/entry/converged_load_balancer

  Regards

  F
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 02, 2010 11:43 AM
    Subject: clustering question


    Hi,

    I searched some articles of FS in a cluster environment
    but there are some unclear writes :

    in http://www.technoclicks.com/article-2545.php from 2006 it says

    "Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

    and this thread from 2008
    http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
    that describes contradiction of how to manage clustering.

    so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

    Regards

    F
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From jonas.gauffin at gmail.com  Tue Mar  2 09:35:33 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 18:35:33 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
Message-ID: <d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>

I solved it by adding '<action application="set" data="sip_h_Diversion="/>'
to my generated dial plan. But why was the header added in the first place?

On Tue, Mar 2, 2010 at 5:20 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I got a incoming call from my provider which goes to a javascript. The
> javascript transfers the call to an external destination (goes back through
> my provider).
> A diversion headers is added by FreeSWITCH on the outbound call. How can I
> remove it (or if not possible, how do I change it)?
>
> javascript:
>         session.setCallerData("caller_id_number", aNumber);
> session.setVariable("gate_caller_id_number", aNumber);
>         session.setVariable("gate_site_id", SITE_ID);
>         session.execute("transfer", this.dtmfBuffer + " XML internal");
>
>
>
> send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200:
>    ------------------------------------------------------------------------
>    INVITE sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>SIP/2.0
>    Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS
>    Max-Forwards: 67
>    From: "0236661XXX" <sip:0243795362 at 212.247.XX.XX>;tag=BHvytKtv47r2a
>    To: <sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>
> >
>    Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5
>    CSeq: 127654862 INVITE
>    Contact: <sip:gw+dalaconnecttele2 at 212.247.XX.XX
> :5060;transport=udp;gw=dalaconnecttele2>
>    User-Agent: Gateon
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 273
>    Diversion: 0 <sip:0 at 212.151.Y.Y
> ;user=phone>;reason=unconditional;counter=1;privacy=off
>    X-FS-Support: update_display
>    Remote-Party-ID: "0236661XXX" <sip:0243795362 at 212.247.XX.XX
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX
>    s=FreeSWITCH
>    c=IN IP4 212.247.XX.XX
>    t=0 0
>    m=audio 32712 RTP/AVP 8 3 101 13
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>
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From msc at freeswitch.org  Tue Mar  2 09:40:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 09:40:56 -0800
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
Message-ID: <87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>

On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> I did make current from my freeswitch src directory, but the system got
> hanged.
> Any one know why this happened.
>

Does it happen every time you make current? If so, does it always happen at
the "expat" lib?
-MC
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From msc at freeswitch.org  Tue Mar  2 09:44:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 09:44:16 -0800
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <251258.58331.qm@web33508.mail.mud.yahoo.com>
References: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
	<251258.58331.qm@web33508.mail.mud.yahoo.com>
Message-ID: <87f2f3b91003020944s77e57eavc5e8e503c0c15cbb@mail.gmail.com>

On Tue, Mar 2, 2010 at 5:24 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi Phillip Jones,
>
> Any concerns, You tell me, I help you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>

Diego, I have a concern: you're supposed to join the FS community conf call
and talk to us about using .NET langs with ESL! :P

Seriously, though, we would love to have you join the conference tomorrow or
any other Wednesday and discuss what's going on with .NET ESL programming.
Moc has a server you can use to share a screen if you'd like.

Thanks!!

-MC
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From brian at freeswitch.org  Tue Mar  2 09:45:27 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 11:45:27 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
Message-ID: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>

Because it should be there if you redirected the call.  

/b

On Mar 2, 2010, at 11:35 AM, Jonas Gauffin wrote:

> I solved it by adding '<action application="set" data="sip_h_Diversion="/>' to my generated dial plan. But why was the header added in the first place?



From brian at freeswitch.org  Tue Mar  2 09:47:39 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 11:47:39 -0600
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
	<87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
Message-ID: <2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>

what os are you on?

/b

On Mar 2, 2010, at 11:40 AM, Michael Collins wrote:

> 
> 
> On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy <lakindia89 at gmail.com> wrote:
> Hi all,
> I did make current from my freeswitch src directory, but the system got hanged.
> Any one know why this happened.
> 
> Does it happen every time you make current? If so, does it always happen at the "expat" lib?
> -MC
> 
> _________

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From sos at sokhapkin.dyndns.org  Tue Mar  2 09:51:48 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 12:51:48 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
Message-ID: <201003021251.48760.sos@sokhapkin.dyndns.org>

FS spends half of CPU here (main thread):

#0  0xffffe424 in __kernel_vsyscall ()
#1  0xb7a82781 in select () from /lib/libc.so.6
#2  0xb7e392eb in apr_sleep () from /opt/freeswitch/lib/libfreeswitch.so.1
#3  0x00000000 in ?? ()

The second half is spent here:

#0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
#1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
#2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8, 
obj=0xb7101da8) at src/switch_loadable_module.c:96
#3  0xb7e379a8 in dummy_worker () from /opt/freeswitch/lib/libfreeswitch.so.1


On Tuesday 02 March 2010, Moises Silva wrote:
> ps -LlFm -p `pidof freeswitch`
>
> That shows you CPU usage per FreeSWITCH thread
>
> Then
>
> pstack `pidof freeswitch`
>
> That shows the stack trace of every FreeSWITCH thread and then you can find
> exactly what is doing.




From anthony.minessale at gmail.com  Tue Mar  2 10:00:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:00:23 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021251.48760.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>

Well, there you go.
your box is having a hard time doing a simple 1ms sleep in a loop for the
centralized core timer.

It's either your kernel or your box.  My box and every box I have do not
have that problem so you may want to consider a new one.

FreeSWITCH is FREE so give it a good home.

The good news is, you can make upwards of 1500 calls on that same 2% of cpu
usage.



On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS spends half of CPU here (main thread):
>
> #0  0xffffe424 in __kernel_vsyscall ()
> #1  0xb7a82781 in select () from /lib/libc.so.6
> #2  0xb7e392eb in apr_sleep () from /opt/freeswitch/lib/libfreeswitch.so.1
> #3  0x00000000 in ?? ()
>
> The second half is spent here:
>
> #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> obj=0xb7101da8) at src/switch_loadable_module.c:96
> #3  0xb7e379a8 in dummy_worker () from
> /opt/freeswitch/lib/libfreeswitch.so.1
>
>
> On Tuesday 02 March 2010, Moises Silva wrote:
> > ps -LlFm -p `pidof freeswitch`
> >
> > That shows you CPU usage per FreeSWITCH thread
> >
> > Then
> >
> > pstack `pidof freeswitch`
> >
> > That shows the stack trace of every FreeSWITCH thread and then you can
> find
> > exactly what is doing.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jonas.gauffin at gmail.com  Tue Mar  2 10:11:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 19:11:54 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
Message-ID: <d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>

Sure I can buy that, but not with a URI that doesnt match the inbound call.

"0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore all
calls through that javascript will be rejected by my provider.


On Tue, Mar 2, 2010 at 6:45 PM, Brian West <brian at freeswitch.org> wrote:

> Because it should be there if you redirected the call.
>
> /b
>
> On Mar 2, 2010, at 11:35 AM, Jonas Gauffin wrote:
>
> > I solved it by adding '<action application="set"
> data="sip_h_Diversion="/>' to my generated dial plan. But why was the header
> added in the first place?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Tue Mar  2 10:16:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 13:16:30 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
Message-ID: <201003021316.30842.sos@sokhapkin.dyndns.org>

I see the same problem on 4 absolutely different boxes with different linux 
kernels.

On Tuesday 02 March 2010, Anthony Minessale wrote:
> Well, there you go.
> your box is having a hard time doing a simple 1ms sleep in a loop for the
> centralized core timer.
>
> It's either your kernel or your box.  My box and every box I have do not
> have that problem so you may want to consider a new one.
>
> FreeSWITCH is FREE so give it a good home.
>
> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
> usage.
>
>
>
> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > FS spends half of CPU here (main thread):
> >
> > #0  0xffffe424 in __kernel_vsyscall ()
> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > #2  0xb7e392eb in apr_sleep () from
> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> >
> > The second half is spent here:
> >
> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > #3  0xb7e379a8 in dummy_worker () from
> > /opt/freeswitch/lib/libfreeswitch.so.1
> >
> > On Tuesday 02 March 2010, Moises Silva wrote:
> > > ps -LlFm -p `pidof freeswitch`
> > >
> > > That shows you CPU usage per FreeSWITCH thread
> > >
> > > Then
> > >
> > > pstack `pidof freeswitch`
> > >
> > > That shows the stack trace of every FreeSWITCH thread and then you can
> >
> > find
> >
> > > exactly what is doing.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From brian at freeswitch.org  Tue Mar  2 10:16:48 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 12:16:48 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
Message-ID: <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>

Was the header there on the inbound invite?

/b

On Mar 2, 2010, at 12:11 PM, Jonas Gauffin wrote:

> Sure I can buy that, but not with a URI that doesnt match the inbound call.
> 
> "0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore all calls through that javascript will be rejected by my provider.
> 
> 

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From gmaruzz at celliax.org  Tue Mar  2 10:23:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Mar 2010 19:23:04 +0100
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021316.30842.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org> 
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com> 
	<201003021316.30842.sos@sokhapkin.dyndns.org>
Message-ID: <7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>

Sergey,

that's not a problem, is how FS works.

That load on the CPU when idle is actually not important. Is just the
minimum, like in a carburettor.

If you put it under load, you will see that there will need some
serious load (many calls) to have it ramp up the cpu usage.

-giovanni

On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> I see the same problem on 4 absolutely different boxes with different linux
> kernels.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
>> Well, there you go.
>> your box is having a hard time doing a simple 1ms sleep in a loop for the
>> centralized core timer.
>>
>> It's either your kernel or your box. ?My box and every box I have do not
>> have that problem so you may want to consider a new one.
>>
>> FreeSWITCH is FREE so give it a good home.
>>
>> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
>> usage.
>>
>>
>>
>> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>> > FS spends half of CPU here (main thread):
>> >
>> > #0 ?0xffffe424 in __kernel_vsyscall ()
>> > #1 ?0xb7a82781 in select () from /lib/libc.so.6
>> > #2 ?0xb7e392eb in apr_sleep () from
>> > /opt/freeswitch/lib/libfreeswitch.so.1 #3 ?0x00000000 in ?? ()
>> >
>> > The second half is spent here:
>> >
>> > #0 ?0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
>> > #1 ?0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
>> > #2 ?0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
>> > obj=0xb7101da8) at src/switch_loadable_module.c:96
>> > #3 ?0xb7e379a8 in dummy_worker () from
>> > /opt/freeswitch/lib/libfreeswitch.so.1
>> >
>> > On Tuesday 02 March 2010, Moises Silva wrote:
>> > > ps -LlFm -p `pidof freeswitch`
>> > >
>> > > That shows you CPU usage per FreeSWITCH thread
>> > >
>> > > Then
>> > >
>> > > pstack `pidof freeswitch`
>> > >
>> > > That shows the stack trace of every FreeSWITCH thread and then you can
>> >
>> > find
>> >
>> > > exactly what is doing.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Tue Mar  2 10:43:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:43:13 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
Message-ID: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>

I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
none of them are Centos 5.3
if you can reproduce it on a box I can visit (Centos 5.3 would be nice), put
in my ssh key and i'll come and look.

Be sure to remember, as you continue to add to this thread that you are
complaining about FS using 2% of your cpu......
That's somewhat amusing to me.


On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Sergey,
>
> that's not a problem, is how FS works.
>
> That load on the CPU when idle is actually not important. Is just the
> minimum, like in a carburettor.
>
> If you put it under load, you will see that there will need some
> serious load (many calls) to have it ramp up the cpu usage.
>
> -giovanni
>
> On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
> > I see the same problem on 4 absolutely different boxes with different
> linux
> > kernels.
> >
> > On Tuesday 02 March 2010, Anthony Minessale wrote:
> >> Well, there you go.
> >> your box is having a hard time doing a simple 1ms sleep in a loop for
> the
> >> centralized core timer.
> >>
> >> It's either your kernel or your box.  My box and every box I have do not
> >> have that problem so you may want to consider a new one.
> >>
> >> FreeSWITCH is FREE so give it a good home.
> >>
> >> The good news is, you can make upwards of 1500 calls on that same 2% of
> cpu
> >> usage.
> >>
> >>
> >>
> >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >> > FS spends half of CPU here (main thread):
> >> >
> >> > #0  0xffffe424 in __kernel_vsyscall ()
> >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> >> > #2  0xb7e392eb in apr_sleep () from
> >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> >> >
> >> > The second half is spent here:
> >> >
> >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> >> > #3  0xb7e379a8 in dummy_worker () from
> >> > /opt/freeswitch/lib/libfreeswitch.so.1
> >> >
> >> > On Tuesday 02 March 2010, Moises Silva wrote:
> >> > > ps -LlFm -p `pidof freeswitch`
> >> > >
> >> > > That shows you CPU usage per FreeSWITCH thread
> >> > >
> >> > > Then
> >> > >
> >> > > pstack `pidof freeswitch`
> >> > >
> >> > > That shows the stack trace of every FreeSWITCH thread and then you
> can
> >> >
> >> > find
> >> >
> >> > > exactly what is doing.
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Tue Mar  2 10:48:34 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 02 Mar 2010 12:48:34 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021316.30842.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>	<201003021251.48760.sos@sokhapkin.dyndns.org>	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
Message-ID: <4B8D5D82.5020304@cartissolutions.com>

On 03/02/2010 12:16 PM, Sergey Okhapkin wrote:
> I see the same problem on 4 absolutely different boxes with different linux
> kernels.
>    

And why do you think this is a problem anyway?  If you want 0% cpu 
utilization, you can always power off the machine.  :-)


> On Tuesday 02 March 2010, Anthony Minessale wrote:
>    
>> Well, there you go.
>> your box is having a hard time doing a simple 1ms sleep in a loop for the
>> centralized core timer.
>>
>> It's either your kernel or your box.  My box and every box I have do not
>> have that problem so you may want to consider a new one.
>>
>> FreeSWITCH is FREE so give it a good home.
>>
>> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
>> usage.
>>      
>
<snip snip>


From andrew at hijacked.us  Tue Mar  2 10:53:02 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 2 Mar 2010 13:53:02 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
Message-ID: <20100302185302.GG1751@hijacked.us>

On Tue, Mar 02, 2010 at 12:43:13PM -0600, Anthony Minessale wrote:
> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> none of them are Centos 5.3
> if you can reproduce it on a box I can visit (Centos 5.3 would be nice), put
> in my ssh key and i'll come and look.
> 
> Be sure to remember, as you continue to add to this thread that you are
> complaining about FS using 2% of your cpu......
> That's somewhat amusing to me.
>

I can confirm this, I just checked 2 of my machines which run identical
hardware and the Ubuntu one has FS sitting about 7% cpu all the time
and the CentOS box has it sitting at 0% (I can't even see it on the
screen). Both machines are idle.

So the moral of the story is that the OS/hardware recommendations are
made for good reason. Disregard them at your own risk.

Andrew


From sos at sokhapkin.dyndns.org  Tue Mar  2 10:53:48 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 13:53:48 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
Message-ID: <201003021353.48590.sos@sokhapkin.dyndns.org>

A program should not be visible in "top" when idle. That's my point.

On Tuesday 02 March 2010, Anthony Minessale wrote:
> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> none of them are Centos 5.3
> if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
> put in my ssh key and i'll come and look.
>
> Be sure to remember, as you continue to add to this thread that you are
> complaining about FS using 2% of your cpu......
> That's somewhat amusing to me.
>
> On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli 
<gmaruzz at celliax.org>wrote:
> > Sergey,
> >
> > that's not a problem, is how FS works.
> >
> > That load on the CPU when idle is actually not important. Is just the
> > minimum, like in a carburettor.
> >
> > If you put it under load, you will see that there will need some
> > serious load (many calls) to have it ramp up the cpu usage.
> >
> > -giovanni
> >
> > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org> wrote:
> > > I see the same problem on 4 absolutely different boxes with different
> >
> > linux
> >
> > > kernels.
> > >
> > > On Tuesday 02 March 2010, Anthony Minessale wrote:
> > >> Well, there you go.
> > >> your box is having a hard time doing a simple 1ms sleep in a loop for
> >
> > the
> >
> > >> centralized core timer.
> > >>
> > >> It's either your kernel or your box.  My box and every box I have do
> > >> not have that problem so you may want to consider a new one.
> > >>
> > >> FreeSWITCH is FREE so give it a good home.
> > >>
> > >> The good news is, you can make upwards of 1500 calls on that same 2%
> > >> of
> >
> > cpu
> >
> > >> usage.
> > >>
> > >>
> > >>
> > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> > >>
> > >> <sos at sokhapkin.dyndns.org>wrote:
> > >> > FS spends half of CPU here (main thread):
> > >> >
> > >> > #0  0xffffe424 in __kernel_vsyscall ()
> > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > >> > #2  0xb7e392eb in apr_sleep () from
> > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> > >> >
> > >> > The second half is spent here:
> > >> >
> > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > >> > #3  0xb7e379a8 in dummy_worker () from
> > >> > /opt/freeswitch/lib/libfreeswitch.so.1
> > >> >
> > >> > On Tuesday 02 March 2010, Moises Silva wrote:
> > >> > > ps -LlFm -p `pidof freeswitch`
> > >> > >
> > >> > > That shows you CPU usage per FreeSWITCH thread
> > >> > >
> > >> > > Then
> > >> > >
> > >> > > pstack `pidof freeswitch`
> > >> > >
> > >> > > That shows the stack trace of every FreeSWITCH thread and then you
> >
> > can
> >
> > >> > find
> > >> >
> > >> > > exactly what is doing.
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From anthony.minessale at gmail.com  Tue Mar  2 11:05:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 13:05:06 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021353.48590.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003021105g51d73cfbs7b88562586366c28@mail.gmail.com>

FS is not really idle when idle.  Calls are not the only thing that cost
cpu.

Its a server daemon.  It has at least 25 threads up including the central
timer thread which is doing an endless-loop of 1ms sleeps.  You have an
active sip engine that still runs in a loop and many other moving parts.







On Tue, Mar 2, 2010 at 12:53 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> A program should not be visible in "top" when idle. That's my point.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
> > I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> > none of them are Centos 5.3
> > if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
> > put in my ssh key and i'll come and look.
> >
> > Be sure to remember, as you continue to add to this thread that you are
> > complaining about FS using 2% of your cpu......
> > That's somewhat amusing to me.
> >
> > On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
> > > Sergey,
> > >
> > > that's not a problem, is how FS works.
> > >
> > > That load on the CPU when idle is actually not important. Is just the
> > > minimum, like in a carburettor.
> > >
> > > If you put it under load, you will see that there will need some
> > > serious load (many calls) to have it ramp up the cpu usage.
> > >
> > > -giovanni
> > >
> > > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org> wrote:
> > > > I see the same problem on 4 absolutely different boxes with different
> > >
> > > linux
> > >
> > > > kernels.
> > > >
> > > > On Tuesday 02 March 2010, Anthony Minessale wrote:
> > > >> Well, there you go.
> > > >> your box is having a hard time doing a simple 1ms sleep in a loop
> for
> > >
> > > the
> > >
> > > >> centralized core timer.
> > > >>
> > > >> It's either your kernel or your box.  My box and every box I have do
> > > >> not have that problem so you may want to consider a new one.
> > > >>
> > > >> FreeSWITCH is FREE so give it a good home.
> > > >>
> > > >> The good news is, you can make upwards of 1500 calls on that same 2%
> > > >> of
> > >
> > > cpu
> > >
> > > >> usage.
> > > >>
> > > >>
> > > >>
> > > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> > > >>
> > > >> <sos at sokhapkin.dyndns.org>wrote:
> > > >> > FS spends half of CPU here (main thread):
> > > >> >
> > > >> > #0  0xffffe424 in __kernel_vsyscall ()
> > > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > > >> > #2  0xb7e392eb in apr_sleep () from
> > > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> > > >> >
> > > >> > The second half is spent here:
> > > >> >
> > > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > > >> > #3  0xb7e379a8 in dummy_worker () from
> > > >> > /opt/freeswitch/lib/libfreeswitch.so.1
> > > >> >
> > > >> > On Tuesday 02 March 2010, Moises Silva wrote:
> > > >> > > ps -LlFm -p `pidof freeswitch`
> > > >> > >
> > > >> > > That shows you CPU usage per FreeSWITCH thread
> > > >> > >
> > > >> > > Then
> > > >> > >
> > > >> > > pstack `pidof freeswitch`
> > > >> > >
> > > >> > > That shows the stack trace of every FreeSWITCH thread and then
> you
> > >
> > > can
> > >
> > > >> > find
> > > >> >
> > > >> > > exactly what is doing.
> > > >> >
> > > >> > _______________________________________________
> > > >> > FreeSWITCH-users mailing list
> > > >> > FreeSWITCH-users at lists.freeswitch.org
> > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> > http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >s http://www.freeswitch.org
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > Cell : +39-347-2665618
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  2 11:05:47 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 14:05:47 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com><191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <1579D06951D94A4DAF34949175410B1F@MOBILEE1705>


----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 02, 2010 1:53 PM
Subject: Re: [Freeswitch-users] What FS is doing when idle?


>A program should not be visible in "top" when idle. That's my point.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
>> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
>> none of them are Centos 5.3
>> if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
>> put in my ssh key and i'll come and look.
>>
>> Be sure to remember, as you continue to add to this thread that you are
>> complaining about FS using 2% of your cpu......
>> That's somewhat amusing to me.
>>
>> On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
>> > Sergey,
>> >
>> > that's not a problem, is how FS works.
>> >
>> > That load on the CPU when idle is actually not important. Is just the
>> > minimum, like in a carburettor.
>> >
>> > If you put it under load, you will see that there will need some
>> > serious load (many calls) to have it ramp up the cpu usage.
>> >
>> > -giovanni
>> >
>> > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
>> >
>> > <sos at sokhapkin.dyndns.org> wrote:
>> > > I see the same problem on 4 absolutely different boxes with different
>> >
>> > linux
>> >
>> > > kernels.
>> > >
>> > > On Tuesday 02 March 2010, Anthony Minessale wrote:
>> > >> Well, there you go.
>> > >> your box is having a hard time doing a simple 1ms sleep in a loop 
>> > >> for
>> >
>> > the
>> >
>> > >> centralized core timer.
>> > >>
>> > >> It's either your kernel or your box.  My box and every box I have do
>> > >> not have that problem so you may want to consider a new one.
>> > >>
>> > >> FreeSWITCH is FREE so give it a good home.
>> > >>
>> > >> The good news is, you can make upwards of 1500 calls on that same 2%
>> > >> of
>> >
>> > cpu
>> >
>> > >> usage.
>> > >>
>> > >>
>> > >>
>> > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>> > >>
>> > >> <sos at sokhapkin.dyndns.org>wrote:
>> > >> > FS spends half of CPU here (main thread):
>> > >> >
>> > >> > #0  0xffffe424 in __kernel_vsyscall ()
>> > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
>> > >> > #2  0xb7e392eb in apr_sleep () from
>> > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
>> > >> >
>> > >> > The second half is spent here:
>> > >> >
>> > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
>> > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
>> > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
>> > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
>> > >> > #3  0xb7e379a8 in dummy_worker () from
>> > >> > /opt/freeswitch/lib/libfreeswitch.so.1
>> > >> >
>> > >> > On Tuesday 02 March 2010, Moises Silva wrote:
>> > >> > > ps -LlFm -p `pidof freeswitch`
>> > >> > >
>> > >> > > That shows you CPU usage per FreeSWITCH thread
>> > >> > >
>> > >> > > Then
>> > >> > >
>> > >> > > pstack `pidof freeswitch`
>> > >> > >
>> > >> > > That shows the stack trace of every FreeSWITCH thread and then 
>> > >> > > you
>> >
>> > can
>> >
>> > >> > find
>> > >> >
>> > >> > > exactly what is doing.
>> > >> >
>> > >> > _______________________________________________
>> > >> > FreeSWITCH-users mailing list
>> > >> > FreeSWITCH-users at lists.freeswitch.org
>> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> > UNSUBSCRIBE:
>> >
>> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > >> > http://www.freeswitch.org
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>> > >s http://www.freeswitch.org
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



I follow this thread, and did top on my Linux Fedora 10 64bits and 
freeswitch last svn version (without mod_vmd that not compiles anymore)
in idle mode and I get

10399 root      20   0  438m  37m 8744 S 11.3  0.5  15:31.61 freeswitch

if I remember the previous version took less resources

Regards

F 



From jonas.gauffin at gmail.com  Tue Mar  2 11:14:57 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 20:14:57 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
Message-ID: <d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>

Thank you! =)
The incoming INVITE had that header. I can't believe that I missed it in the
first place.

On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:

> Was the header there on the inbound invite?
>
> /b
>
> On Mar 2, 2010, at 12:11 PM, Jonas Gauffin wrote:
>
> Sure I can buy that, but not with a URI that doesnt match the inbound call.
>
> "0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore
> all calls through that javascript will be rejected by my provider.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Tue Mar  2 11:15:14 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Mar 2010 13:15:14 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>

For whatever it's worth; I have FS running on a 1.6MHz processor,
CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.

On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No single
> SIP request hits it. What does it doing? I unloaded all modules which can be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux ?2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686 Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Mar  2 11:25:29 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:25:29 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
	<d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
Message-ID: <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>

And are you using Proxy Media?

/b

On Mar 2, 2010, at 1:14 PM, Jonas Gauffin wrote:

> Thank you! =)
> The incoming INVITE had that header. I can't believe that I missed it in the first place.
> 
> On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:
> Was the header there on the inbound invite?
> 
> /b

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From brian at freeswitch.org  Tue Mar  2 11:25:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:25:10 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
Message-ID: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>

How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

:P

/b

On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

> For whatever it's worth; I have FS running on a 1.6MHz processor,
> CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.



From lists at redbonez.net  Tue Mar  2 11:27:45 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 2 Mar 2010 12:27:45 -0700
Subject: [Freeswitch-users] xml_cdr http post differentiation between legs
Message-ID: <034401caba3e$70bbb950$52332bf0$@net>

Is there any way to tell whether it is the a-leg or b-leg when receiving the
POST data from mod_xml_cdr?

 

I wrote a parser in Python which parses the a_*.cdr.xml log files, and only
grabs the info in the associated b-leg log files if it needs additional
information(for example - in the case of a call that ends with mod_fifo).  I
want to adapt this to using the POST features of mod_xml_cdr so that it can
be real time. 

 

Any suggestions are greatly appreciated,

-Adam 

 

 

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From brian at freeswitch.org  Tue Mar  2 11:30:36 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:30:36 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
Message-ID: <38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>

I need you to email me your username and password for this challenge so I can calc this to verify please.

/b

On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:

> You are both correct. 
> 
> Issue created.
> 
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
> Jonas,
> 	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.
> 
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
> 
> Thanks,
> Brian
> 
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
> 
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>> 
>> 
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
>> Hello,
>> 
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
>> 
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>> 
>> 
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>> 
>> Regards,
>>   Jonas
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar  2 11:41:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:41:06 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
Message-ID: <1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>

What really really concerns me about this is the fact this is MD5 digest auth.

Neither of these:
4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47

Are NOT the right length to be an MD5 of any auth data which would be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really does concern me.

Anyone else have input on this one?

/b



On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:

> You are both correct. 
> 
> Issue created.
> 
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
> Jonas,
> 	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.
> 
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
> 
> Thanks,
> Brian
> 
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
> 
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>> 
>> 
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
>> Hello,
>> 
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
>> 
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>> 
>> 
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>> 
>> Regards,
>>   Jonas
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chris.chen2004 at gmail.com  Tue Mar  2 12:08:11 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 2 Mar 2010 15:08:11 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
Message-ID: <507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>

I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is
common to net books from ASUS and like, I have one running on EEE box N270.

Chris

On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> How many calls can that 1.6MHz CPU do?  :P And how long does it take to
> compile?
>
> :P
>
> /b
>
> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>
> > For whatever it's worth; I have FS running on a 1.6MHz processor,
> > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Tue Mar  2 12:09:56 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Mar 2010 14:09:56 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
Message-ID: <c55d46951003021209w45c35a9dw57d97f219236d09@mail.gmail.com>

Next time i compile I will let you know. But if I remember correctly,
it takes some few hours. And as far as calls are concerned I don't
intend  to have more than 8 calls at a time. So the question is really
irrelevant for my settings ;)

Max.

On Tue, Mar 2, 2010 at 1:25 PM, Brian West <brian at freeswitch.org> wrote:
> How many calls can that 1.6MHz CPU do? ?:P And how long does it take to compile?
>
> :P
>
> /b
>
> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>
>> For whatever it's worth; I have FS running on a 1.6MHz processor,
>> CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From infos at madovsky.org  Tue Mar  2 12:19:31 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 15:19:31 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com><744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
Message-ID: <91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>

I tried to downgrade FS to version 16597, all compiles well (even mod_vmd)
so with exactly the same config as last svn trunk I get with -vm :

21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch -> CPU = 2.3%

and without -vm
25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch -> CPU 4.00%

which is less as the last version that takes almost 12%.

hope this helps

Regards

F





  ----- Original Message ----- 
  From: Chris Chen 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 3:08 PM
  Subject: Re: [Freeswitch-users] What FS is doing when idle?


  I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.

  Chris 


  On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

    How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

    :P

    /b


    On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

    > For whatever it's worth; I have FS running on a 1.6MHz processor,
    > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.



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





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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Tue Mar  2 12:22:23 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:22:23 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
Message-ID: <25008BD4-47DE-46BF-ABB9-777467E5E80C@freeswitch.org>

I was just poking some fun... I knew he was on a 1.6ghz :P

/b

On Mar 2, 2010, at 2:08 PM, Chris Chen wrote:

> I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.
> 
> Chris 



From jonas.gauffin at gmail.com  Tue Mar  2 12:27:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:27:54 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
	<d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
	<1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>
Message-ID: <d021d3be1003021227r3eb9cbc6md5f337eba7d5ede@mail.gmail.com>

No. I'm only using proxy_media on fax devices.

On Tue, Mar 2, 2010 at 8:25 PM, Brian West <brian at freeswitch.org> wrote:

> And are you using Proxy Media?
>
> /b
>
> On Mar 2, 2010, at 1:14 PM, Jonas Gauffin wrote:
>
> Thank you! =)
> The incoming INVITE had that header. I can't believe that I missed it in
> the first place.
>
> On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Was the header there on the inbound invite?
>>
>> /b
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jonas.gauffin at gmail.com  Tue Mar  2 12:37:27 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:37:27 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
Message-ID: <d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>

Even if the string is incorrect, shouldn't FreeSWITCH use it without
modification in it's calculations (and return the same nonce in the
response)?

On Tue, Mar 2, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:

> What really really concerns me about this is the fact this is MD5 digest
> auth.
>
> Neither of these:
> 4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
> 4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47
>
> Are NOT the right length to be an MD5 of any auth data which would
> be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really
> does concern me.
>
> Anyone else have input on this one?
>
> /b
>
>
>
> On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:
>
> You are both correct.
>
> Issue created.
>
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Jonas,
>> Also I need a complete pcap of this without modifications.  I can't hand
>> calculate this to see what is going on without intact data. Please open a
>> jira and email me the pcap in private if you must.
>>
>> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>>
>> Thanks,
>> Brian
>>
>> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>>
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>>
>>
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I got a problem with rejected calls due to 403 (happens sometimes).
>>> My sip provider found the problem, the nonce used by FS is sometimes
>>> incorrect (I got the trace from my sip provider).
>>>
>>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>>
>>>
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>>> sofia/external/0707728XXX SOFIA INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>>> sofia/external/0707728XXX SOFIA ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [terminated][407]
>>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar  2 12:37:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 14:37:28 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
	<91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
Message-ID: <191c3a031003021237t30f46edp2978f258c54c326b@mail.gmail.com>

and make calls with both and measure the jitter with wireshark.


On Tue, Mar 2, 2010 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:

>  I tried to downgrade FS to version 16597, all compiles well (even
> mod_vmd)
> so with exactly the same config as last svn trunk I get with -vm :
>
> 21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch ->
> CPU = 2.3%
>
> and without -vm
> 25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch ->
> CPU 4.00%
>
> which is less as the last version that takes almost 12%.
>
> hope this helps
>
> Regards
>
> F
>
>
>
>
>  ----- Original Message -----
> *From:* Chris Chen <chris.chen2004 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 02, 2010 3:08 PM
> *Subject:* Re: [Freeswitch-users] What FS is doing when idle?
>
> I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is
> common to net books from ASUS and like, I have one running on EEE box N270.
>
> Chris
>
> On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:
>
>> How many calls can that 1.6MHz CPU do?  :P And how long does it take to
>> compile?
>>
>> :P
>>
>> /b
>>
>> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>>
>> > For whatever it's worth; I have FS running on a 1.6MHz processor,
>> > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar  2 12:44:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:44:31 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
	<d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>
Message-ID: <F4B955FF-0B87-4CDC-92D4-04832A5004BB@freeswitch.org>

Not 100% sure since I don't have the details of your sip account to test this... I need to try this in eyebeam or something to see how it behaves.

/b

On Mar 2, 2010, at 2:37 PM, Jonas Gauffin wrote:

> Even if the string is incorrect, shouldn't FreeSWITCH use it without modification in it's calculations (and return the same nonce in the response)?
> 
> On Tue, Mar 2, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> What really really concerns me about this is the fact this is MD5 digest auth.
> 
> Neither of these:
> 4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
> 4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47
> 
> Are NOT the right length to be an MD5 of any auth data which would be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really does concern me.
> 
> Anyone else have input on this one?
> 
> /b

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From msc at freeswitch.org  Tue Mar  2 12:44:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:44:34 -0800
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021353.48590.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>

On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> A program should not be visible in "top" when idle. That's my point.
>

That's a dangerous assumption. Do really know for a certainty that a program
"should not" be visible in top when it's "idle"? Like Tony said below,
"idle" is misleading. FreeSWITCH is working its but off when you think it's
"idle." Make 100 simultaneous calls and see what the CPU usage is. I think
you'll be happy with the results.

For the record, we are going to start a wiki page that explains all of this.
It's kind of analogous to the whole "Linux eats all my RAM" discussion. We
should have something documented in a few days.
-MC
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From msc at freeswitch.org  Tue Mar  2 12:48:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:48:30 -0800
Subject: [Freeswitch-users] wiki broken link pocketsphinx
In-Reply-To: <30970655E0CD476A95CF8382A2742084@MOBILEE1705>
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
	<5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>
	<30970655E0CD476A95CF8382A2742084@MOBILEE1705>
Message-ID: <87f2f3b91003021248s1f525a28xec3daefe20b9cd7c@mail.gmail.com>

On Tue, Mar 2, 2010 at 7:09 AM, Madovsky <infos at madovsky.org> wrote:

>  apparently Diego updated it.
> I don't know if I'm skilled enough to elaborate
>
>
AFAICT the link is correct now.
-MC
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From brian at freeswitch.org  Tue Mar  2 12:54:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:54:31 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <F58AF730-64DE-429A-A698-4DE13CC743FE@freeswitch.org>

Also add "Why is my load average so high with FreeSWITCH?" to that list of things we need to explain.  Because load average is just a metric of how many threads are in the run queue.. its really no way to tell how busy a box really is since most threads have narcolepsy.

/b

On Mar 2, 2010, at 2:44 PM, Michael Collins wrote:

> For the record, we are going to start a wiki page that explains all of this. It's kind of analogous to the whole "Linux eats all my RAM" discussion. We should have something documented in a few days.
> -MC



From jonas.gauffin at gmail.com  Tue Mar  2 12:55:28 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:55:28 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
Message-ID: <d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>

Ok. I've done a digest calculation and got a match.
The "response" value is calculated using the nonce that FS sends in the
INVITE, and not the nonce that is recieved in the 407 reply from my
provider.


On Tue, Mar 2, 2010 at 8:30 PM, Brian West <brian at freeswitch.org> wrote:

> I need you to email me your username and password for this challenge so I
> can calc this to verify please.
>
> /b
>
> On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:
>
> You are both correct.
>
> Issue created.
>
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Jonas,
>> Also I need a complete pcap of this without modifications.  I can't hand
>> calculate this to see what is going on without intact data. Please open a
>> jira and email me the pcap in private if you must.
>>
>> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>>
>> Thanks,
>> Brian
>>
>> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>>
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>>
>>
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I got a problem with rejected calls due to 403 (happens sometimes).
>>> My sip provider found the problem, the nonce used by FS is sometimes
>>> incorrect (I got the trace from my sip provider).
>>>
>>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>>
>>>
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>>> sofia/external/0707728XXX SOFIA INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>>> sofia/external/0707728XXX SOFIA ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [terminated][407]
>>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar  2 12:55:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:55:38 -0800
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8D2BA6.1030907@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
	<FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
	<4B8D2BA6.1030907@cachecomm.com>
Message-ID: <87f2f3b91003021255m1b27caf3gc3ef95645e4162ab@mail.gmail.com>

I removed the Netgear P.O.S . and put the phones behind a linux firewall all
is good.

Let's put this text in neon letters ten feet high!
-MC
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From brian at freeswitch.org  Tue Mar  2 12:59:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:59:10 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
	<d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>
Message-ID: <1C428CC2-8166-4061-9378-9FEC0388EDB6@freeswitch.org>

Please include all these details in the jira.

/b

On Mar 2, 2010, at 2:55 PM, Jonas Gauffin wrote:

> Ok. I've done a digest calculation and got a match.
> The "response" value is calculated using the nonce that FS sends in the INVITE, and not the nonce that is recieved in the 407 reply from my provider.
> 
> 
> On Tue, Mar 2, 2010 at 8:30 PM, Brian West <brian at freeswitch.org> wrote:
> I need you to email me your username and password for this challenge so I can calc this to verify please.
> 
> /b

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From anthony.minessale at gmail.com  Tue Mar  2 13:05:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 15:05:04 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <191c3a031003021305k7de4b3f8ie6bebebd19ceb8a3@mail.gmail.com>

Since this topic is philosophical to begin with, "What does a FS box do when
idle, and if a tree fell in the woods while a FS box was in a nearby cabin
would a bear shit in the woods....."

Another philosophical question would be, If a mailing-list thread persisted
for more than 6 hours would it take more or less time to just change the
code to make people stop complaining about free software  than it does to
politely not concern one's self with the issue.  I chose the first option,
you can update to trunk now and probably see your precious 0.0% so you can
get absolutely no use from your computer.  As a side note, my change
effectively proves  that the performance of pthread's mutexing and
conditional broadcast implementation has gone downhill since the version on
CentOS.


On Tue, Mar 2, 2010 at 2:44 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> A program should not be visible in "top" when idle. That's my point.
>>
>
> That's a dangerous assumption. Do really know for a certainty that a
> program "should not" be visible in top when it's "idle"? Like Tony said
> below, "idle" is misleading. FreeSWITCH is working its but off when you
> think it's "idle." Make 100 simultaneous calls and see what the CPU usage
> is. I think you'll be happy with the results.
>
> For the record, we are going to start a wiki page that explains all of
> this. It's kind of analogous to the whole "Linux eats all my RAM"
> discussion. We should have something documented in a few days.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From phunk0000 at hotmail.com  Tue Mar  2 13:15:06 2010
From: phunk0000 at hotmail.com (Todd)
Date: Tue, 2 Mar 2010 16:15:06 -0500
Subject: [Freeswitch-users] Play message upon call anwser
Message-ID: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks 

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From freeswitch at cartissolutions.com  Tue Mar  2 13:27:50 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 02 Mar 2010 15:27:50 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <4B8D82D6.90007@cartissolutions.com>

On 03/02/2010 02:44 PM, Michael Collins wrote:
>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin 
> <sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>> wrote:
>
>     A program should not be visible in "top" when idle. That's my point.
>
>
> That's a dangerous assumption. Do really know for a certainty that a 
> program "should not" be visible in top when it's "idle"? Like Tony 
> said below, "idle" is misleading. FreeSWITCH is working its but off 
> when you think it's "idle." Make 100 simultaneous calls and see what 
> the CPU usage is. I think you'll be happy with the results.
>
<snip snip>

Cue up applicable car analogy....

I ask Sergey this question:  When you turn on your car and let it while 
in either a parking gear or in neutral, your car is considered 
"idling".  But does the car still not use oxygen and gasoline, and 
produce heat and waste products all the while idling?  That is how you 
can think of a complex system such as FreeSWITCH.  It is a machine with 
a lot of parts, and idling means that it is in a "ready" state, but does 
not mean that it is "turned off".

-Yossi

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From robert.hadley at teotech.com  Tue Mar  2 13:31:24 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Mar 2010 13:31:24 -0800
Subject: [Freeswitch-users] Delete custom voicemail greetings
Message-ID: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>

Hi FS users,

 

How do I delete custom voicemail greetings and go back to the default
greeting? 

 

Thanks,

Robert

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From pjintheusa at gmail.com  Tue Mar  2 13:40:30 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 2 Mar 2010 16:40:30 -0500
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <251258.58331.qm@web33508.mail.mud.yahoo.com>
References: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
	<251258.58331.qm@web33508.mail.mud.yahoo.com>
Message-ID: <367751821003021340m4708598bu6c142fd74a0757d@mail.gmail.com>

Thank you Diego - I will be taking a look later this week and will let you
know how it goes.

Or see you on the conf call tomorrow!

On Tue, Mar 2, 2010 at 8:24 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi Phillip Jones,
>
> Any concerns, You tell me, I help you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 3/1/10, Phillip Jones <pjintheusa at gmail.com> wrote:
>
> > From: Phillip Jones <pjintheusa at gmail.com>
> > Subject: Re: [Freeswitch-users] .NET / ESL
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Monday, March 1, 2010, 10:01 PM
> > Thanks
> >
> > On Mon, Mar 1, 2010 at 9:42 PM,
> > Brian West <brian at freeswitch.org>
> > wrote:
> >
> > http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
> >
> >
> >
> > in the managed folder.
> >
> >
> >
> > /b
> >
> >
> >
> > On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
> >
> >
> >
> > > Hi there,
> >
> > >
> >
> > > I am sure that I have read somewhere that someone is
> > developing a new .NET library for FreeSWITCH ESL. If so can
> > someone point me to some info / code etc. so I can take a
> > look.
> >
> > >
> >
> > > Thanks
> >
> > >
> >
> > > pj
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar  2 17:01:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 17:01:41 -0800
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <4B8D82D6.90007@cartissolutions.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
	<4B8D82D6.90007@cartissolutions.com>
Message-ID: <87f2f3b91003021701i6848c9e9hfb665ee0f1892802@mail.gmail.com>

It's sad that my computer uses more than 2% CPU whenever I try to open this
thread in my email client...
-MC

On Tue, Mar 2, 2010 at 1:27 PM, Yossi Neiman <freeswitch at cartissolutions.com
> wrote:

>  On 03/02/2010 02:44 PM, Michael Collins wrote:
>
>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> A program should not be visible in "top" when idle. That's my point.
>>
>
> That's a dangerous assumption. Do really know for a certainty that a
> program "should not" be visible in top when it's "idle"? Like Tony said
> below, "idle" is misleading. FreeSWITCH is working its but off when you
> think it's "idle." Make 100 simultaneous calls and see what the CPU usage
> is. I think you'll be happy with the results.
>
>   <snip snip>
>
> Cue up applicable car analogy....
>
> I ask Sergey this question:  When you turn on your car and let it while in
> either a parking gear or in neutral, your car is considered "idling".  But
> does the car still not use oxygen and gasoline, and produce heat and waste
> products all the while idling?  That is how you can think of a complex
> system such as FreeSWITCH.  It is a machine with a lot of parts, and idling
> means that it is in a "ready" state, but does not mean that it is "turned
> off".
>
> -Yossi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Mar  2 17:26:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 09:26:52 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
Message-ID: <23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>

Thanks, I removed @30i and it works, however, CELT, doesn't.

CODECS IN               PCMU,PCMA,iLBC,CELT,GSM
CODECS OUT              PCMU,PCMA,iLBC,CELT,GSM


2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[GSM:3:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[GSM:3:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[GSM:3:8000:20]

CELT at 32000h

2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[GSM:3:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[GSM:3:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[GSM:3:8000:20]

2010/3/2 Brian West <brian at freeswitch.org>:
> Yes that is because its @30i, All codecs offered MUST be 30ms if you want to offer iLBC at 30i. ?So add @30i to all but GSM since you can't do GSM @ 30.
>
> /b
> On Mar 2, 2010, at 1:01 AM, Seven Du wrote:
>
>> sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)
>>
>>
>> 2010/3/2 Seven Du <dujinfang at gmail.com>:
>>> I have iLBC configured in profile internal
>>>
>>> CODECS IN ? ? ? ? ? ? ? PCMU,PCMA,iLBC at 30i,GSM
>>> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,iLBC at 30i,GSM
>>>
>>> and ? ? <param name="inbound-codec-negotiation" value="generous"/>
>>>
>>> but it seems not comparing with iLBC
>>>
>>> ?010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
>>> payload to 101
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>>>
>>>
>>> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>>>
>>> Thanks.
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Mar  2 17:45:30 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 19:45:30 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable  destination
In-Reply-To: <23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
Message-ID: <AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>

Celt can only do 10ms :P

/b

On Mar 2, 2010, at 7:26 PM, Seven Du wrote:

> Thanks, I removed @30i and it works, however, CELT, doesn't.



From infos at madovsky.org  Tue Mar  2 18:47:25 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 21:47:25 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>
Message-ID: <7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>

make install samples
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 4:31 PM
  Subject: [Freeswitch-users] Delete custom voicemail greetings


  Hi FS users,

   

  How do I delete custom voicemail greetings and go back to the default greeting? 

   

  Thanks,

  Robert



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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Mar  2 18:56:57 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 21:56:57 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com><744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org><507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com><91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
	<191c3a031003021237t30f46edp2978f258c54c326b@mail.gmail.com>
Message-ID: <509D15C949DC42898735CB1CC3CDD7E4@MOBILEE1705>

Ok I will do it, but I  can make only one call since I work alone on FS :D
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 3:37 PM
  Subject: Re: [Freeswitch-users] What FS is doing when idle?


  and make calls with both and measure the jitter with wireshark.



  On Tue, Mar 2, 2010 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:

    I tried to downgrade FS to version 16597, all compiles well (even mod_vmd)
    so with exactly the same config as last svn trunk I get with -vm :

    21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch -> CPU = 2.3%

    and without -vm
    25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch -> CPU 4.00%

    which is less as the last version that takes almost 12%.

    hope this helps

    Regards

    F





      ----- Original Message ----- 
      From: Chris Chen 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 02, 2010 3:08 PM
      Subject: Re: [Freeswitch-users] What FS is doing when idle?


      I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.

      Chris 


      On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

        How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

        :P

        /b


        On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

        > For whatever it's worth; I have FS running on a 1.6MHz processor,
        > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.



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





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


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


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





  -- 
  Anthony Minessale II

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

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

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



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From dujinfang at gmail.com  Tue Mar  2 19:45:15 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 11:45:15 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
Message-ID: <23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>

Oh, yes, but then how I set the pref codec string? I want to also
support other codecs and other ms, now I set to

 PCMU,PCMA,iLBC,CELT,GSM

with the above string I can do 30ms iLBC


2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[iLBC:98:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3580 Substituting
codec iLBC at 30i@8000h
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:2333 Set Codec
sofia/internal/1001 at 114.80.200.174 iLBC/8000 30 ms 240 samples


should I use another profile that only do CELT with 10ms?

2010/3/3 Brian West <brian at freeswitch.org>:
> Celt can only do 10ms :P
>
> /b
>
> On Mar 2, 2010, at 7:26 PM, Seven Du wrote:
>
>> Thanks, I removed @30i and it works, however, CELT, doesn't.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Mar  2 19:51:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 21:51:38 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable  destination
In-Reply-To: <23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
	<23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
Message-ID: <D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>

Enable late negotiation? And pick after the fact?

/b

On Mar 2, 2010, at 9:45 PM, Seven Du wrote:

> Oh, yes, but then how I set the pref codec string? I want to also
> support other codecs and other ms, now I set to
> 
> PCMU,PCMA,iLBC,CELT,GSM
> 
> with the above string I can do 30ms iLBC



From msc at freeswitch.org  Tue Mar  2 20:03:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 20:03:53 -0800
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
Message-ID: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

>  Hey List..  I am trying to setup my dial plan so whenever a call is
> answered a recording is played to the person who answered the phone before
> the call is connected?.  Any help would be great.   Thanks
>
Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC
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From dujinfang at gmail.com  Tue Mar  2 20:20:20 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 12:20:20 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
	<23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
	<D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>
Message-ID: <23f91031003022020o24c809afs55afd9574afa4ee5@mail.gmail.com>

2010/3/3 Brian West <brian at freeswitch.org>:
> Enable late negotiation? And pick after the fact?

Understood. Is that the conf of freeswitch conference call?

But I think it would be simpler to start a new profile in my scenario.

And, doesn't it make sense to also compare mixed ms?


Thanks.
>
> /b
>
> On Mar 2, 2010, at 9:45 PM, Seven Du wrote:
>
>> Oh, yes, but then how I set the pref codec string? I want to also
>> support other codecs and other ms, now I set to
>>
>> PCMU,PCMA,iLBC,CELT,GSM
>>
>> with the above string I can do 30ms iLBC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lakindia89 at gmail.com  Tue Mar  2 20:48:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 3 Mar 2010 10:18:29 +0530
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
	<87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
	<2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>
Message-ID: <7d79b3931003022048y48c8ecbdrc1c0e72877a9f6c4@mail.gmail.com>

Hey! when I did the make current for several times yesterday, the system got
hanged, and on the same line only.

But today when I tried, it went fine. I've not done any changes other than
shutting the system and started it again..

This time I got revision 16873 with openap(1048).
But yesterday it was 16859 with openzap(1047) and I just have a though that
it may be because of this.
Thank u all.


On Tue, Mar 2, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:

> what os are you on?
>
> /b
>
> On Mar 2, 2010, at 11:40 AM, Michael Collins wrote:
>
>
>
> On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>> I did make current from my freeswitch src directory, but the system got
>> hanged.
>> Any one know why this happened.
>>
>
> Does it happen every time you make current? If so, does it always happen at
> the "expat" lib?
> -MC
>
> _________
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Tue Mar  2 21:04:46 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 2 Mar 2010 23:04:46 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
Message-ID: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>

Hi all,

Is anyone running FreeSWITCH virtualized (xen or vmware) in  
production?  I saw ec2 and xen mentioned a few places.  If so, are  
there tips or tricks for handle timer issues which affect virtual  
machines in general, such as recommended divider or clocksource  
settings?  I'm seeing funny timing issues while testing with audio  
recording and playback that speed up or slow down.

I have tried both paravirt and hvm.  The default xen kernel has a  
timer of 250, so I tried to hvm instead using the stock 5.3 kernel  
with the 1khz timer but didn't see much improvement.

All of this was tested on 64-bit CentOS 5.3.
- Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1  
SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP  
Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan  
21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=1000


I'm sure physical hosts are the de-facto best practice, but are  
virtual machines not worth it at all?

Cheers,
Rob


From brian at microcomaustralia.com.au  Tue Mar  2 21:27:29 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 3 Mar 2010 16:27:29 +1100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
Message-ID: <3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>

On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production? ?I saw ec2 and xen mentioned a few places. ?If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings? ?I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.

I tried tdm400+xen DOM0+asterisk years ago, on a single processor
machine but gave up. zttest returned good results, but analogue
quality has intermittently very poor quality and it made telephone
conversations difficult to understand. Especially a problem when
hourly cron jobs would start on all VMs at exactly the same time, but
was a problem at other times too. Plus the fact people who used the
phone more often then me didn't report the problem, so I thought it
was fine.

Plus the general fact I didn't really like the idea of running
Asterisk on DOM0, and running it on DOMU seemed to imply even more
overheard, as DOM0 would have to process the interrupts and pass them
to the DOMU.

Unfortunately I don't remember what scheduler I used, maybe sedf?

I suspect a multi-processor system would be a lot better suited for
the task. Also might be OK if you don't use zaptel hardware... I seem
to remember these cards generate a huge number of interrupts.

While this was with Asterisk, I suspect my results were not specific
to Asterisk in anyway.
-- 
Brian May <brian at microcomaustralia.com.au>


From jmesquita at freeswitch.org  Tue Mar  2 22:00:11 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 3 Mar 2010 03:00:11 -0300
Subject: [Freeswitch-users] FScomm
In-Reply-To: <D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
	<D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
Message-ID: <ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>

This is definitely a problem with your Qt installation. You need to be able
to find the libs to link it, otherwise, unresolved symbols everywhere!

Regards,
Jo?o Mesquita
FSComm Developer


On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:

>
> ----- Original Message -----
> From: "Michael Jerris" <mike at jerris.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 24, 2010 3:40 AM
> Subject: Re: [Freeswitch-users] FScomm
>
>
>
> On Feb 23, 2010, at 6:18 PM, Madovsky wrote:
>
> >
> > ----- Original Message -----
> > From: Jeff Lenk
> > To: freeswitch-users at lists.freeswitch.org
> > Sent: Tuesday, February 23, 2010 5:05 PM
> > Subject: Re: [Freeswitch-users] FScomm
> >
> >
> > http://wiki.freeswitch.org/wiki/FSComm#Linux
> >
> > you must run those from the FSComm directory
> >
>
> ?.
>
> >
> > It's what I did,
> > but from FS trunk, inside fscomm directory,
> > there s only
> >
> > account.cpp  conf                fshost.h        mainwindow.ui
> > resources.qrc
> > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> > call.cpp     FSComm.pro          mainwindow.cpp  preferences
> > call.h       fshost.cpp          mainwindow.h    resources
> >
>
>
> Read those installation instructions again and do them step by step, you
> skipped one.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Ok now I have
>
> [root at node250 fscomm]# qmake
> WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
> in SOURCES
> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
> HEADERS
> WARNING: Found potential symbol conflict of prefdialog.cpp
> (preferences/prefdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of prefdialog.h
> (preferences/prefdialog.h) in HEADERS
> WARNING: Found potential symbol conflict of accountdialog.cpp
> (preferences/accountdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of accountdialog.h
> (preferences/accountdialog.h) in HEADERS
>
> [root at node250 fscomm]# qmake
> WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
> in SOURCES
> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
> HEADERS
> WARNING: Found potential symbol conflict of prefdialog.cpp
> (preferences/prefdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of prefdialog.h
> (preferences/prefdialog.h) in HEADERS
> WARNING: Found potential symbol conflict of accountdialog.cpp
> (preferences/accountdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of accountdialog.h
> (preferences/accountdialog.h) in HEADERS
> [root at node250 fscomm]# make
> Makefile:278: warning: overriding commands for target `prefdialog.o'
> Makefile:215: warning: ignoring old commands for target `prefdialog.o'
> Makefile:285: warning: overriding commands for target `accountdialog.o'
> Makefile:234: warning: ignoring old commands for target `accountdialog.o'
> Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
> Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
> Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
> Makefile:307: warning: ignoring old commands for target
> `moc_accountdialog.o'
> Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
> Makefile:326: warning: ignoring old commands for target
> `moc_mainwindow.cpp'
> Makefile:350: warning: overriding commands for target
> `preferences/moc_prefdialog.cpp'
> Makefile:332: warning: ignoring old commands for target
> `preferences/moc_prefdialog.cpp'
> Makefile:353: warning: overriding commands for target
> `preferences/moc_accountdialog.cpp'
> Makefile:341: warning: ignoring old commands for target
> `preferences/moc_accountdialog.cpp'
> g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
> -fexceptions
>  -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic
>  -DQT_NO_DEBUG
>  -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT
> -I/usr/lib64/qt-3.3/mkspecs/default
>  -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src
> -I/usr/lib64/qt-3.3/include
>  -o main.o main.cpp
> main.cpp:31:25: error: QSplashScreen: No such file or directory
> In file included from main.cpp:32:
> mainwindow.h:34:23: error: QMainWindow: No such file or directory
> mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
> mainwindow.h:36:25: error: QSignalMapper: No such file or directory
> mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
> In file included from mainwindow.h:39,
>                 from main.cpp:32:
> ./fshost.h:32:19: error: QThread: No such file or directory
> ./fshost.h:33:17: error: QHash: No such file or directory
> ./fshost.h:34:26: error: QSharedPointer: No such file or directory
> In file included from ./fshost.h:36,
>                 from mainwindow.h:39,
>                 from main.cpp:32:
> ./call.h:32:18: error: QtCore: No such file or directory
> ./call.h:33:19: error: QString: No such file or directory
> In file included from mainwindow.h:42,
>                 from main.cpp:32:
> preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
> preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
> directory
> preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
> In file included from ./fshost.h:37,
>                 from mainwindow.h:39,
>                 from main.cpp:32:
> ./account.h:18: error: expected constructor, destructor, or type conversion
> before ?static?
> In file included from mainwindow.h:39,
>                 from main.cpp:32:
> ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
> /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
> QThread?
> ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:46: error: expected ?;? before ?<? token
> ./fshost.h:47: error: expected `;' before ?QSharedPointer?
> ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:47: error: expected ?;? before ?<? token
> ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:48: error: template argument 1 is invalid
> ./fshost.h:48: error: expected unqualified-id before ?>? token
> ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:49: error: expected ?;? before ?<? token
> ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:50: error: expected ?;? before ?<? token
> ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:51: error: expected ?;? before ?<? token
> ./fshost.h:52: error: ?QSharedPointer? has not been declared
> ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:60: error: ?QSharedPointer? has not been declared
> ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:61: error: ?QSharedPointer? has not been declared
> ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:62: error: ?QSharedPointer? has not been declared
> ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:63: error: ?QSharedPointer? has not been declared
> ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:64: error: ?QSharedPointer? has not been declared
> ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:65: error: ?QSharedPointer? has not been declared
> ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:66: error: ?QSharedPointer? has not been declared
> ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:67: error: ?QSharedPointer? has not been declared
> ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:71: error: ?QSharedPointer? has not been declared
> ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:78: error: template argument 2 is invalid
> ./fshost.h:78: error: expected unqualified-id before ?>? token
> ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:79: error: template argument 2 is invalid
> ./fshost.h:79: error: expected unqualified-id before ?>? token
> ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
> In file included from mainwindow.h:42,
>                 from main.cpp:32:
> preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
> QDialog?
> /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
> QDialog?
> preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
> ?QSettings? with no type
> preferences/prefdialog.h:31: error: expected ?;? before ?*? token
> In file included from main.cpp:32:
> mainwindow.h:48: error: expected class-name before ?{? token
> mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
> mainwindow.h:71: error: ?QSharedPointer? has not been declared
> mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:72: error: ?QSharedPointer? has not been declared
> mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:73: error: ?QSharedPointer? has not been declared
> mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:74: error: ?QSharedPointer? has not been declared
> mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:75: error: ?QSharedPointer? has not been declared
> mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:78: error: ?QSharedPointer? has not been declared
> mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:79: error: ?QSharedPointer? has not been declared
> mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:80: error: ?QSharedPointer? has not been declared
> mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
> mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
> mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
> no type
> mainwindow.h:86: error: expected ?;? before ?*? token
> mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
> with no type
> mainwindow.h:88: error: expected ?;? before ?*? token
> main.cpp: In function ?int main(int, char**)?:
> main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
> type
> main.cpp:42: error: ?QSplashScreen? was not declared in this scope
> main.cpp:42: error: ?splash? was not declared in this scope
> main.cpp:42: error: expected type-specifier before ?QSplashScreen?
> main.cpp:42: error: expected `;' before ?QSplashScreen?
> main.cpp:48: error: no matching function for call to
> ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
> /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
> QObject::connect(const QObject*, const char*, const QObject*, const char*,
> Qt::ConnectionType)
> /usr/include/QtCore/qobject.h:307: note:                 bool
> QObject::connect(const QObject*, const char*, const char*,
> Qt::ConnectionType) const
> main.cpp:49: error: ?class FSHost? has no member named ?start?
> main.cpp:41: warning: unused variable ?image?
> ./fshost.h: At global scope:
> ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)?
> defined
> but not used
> make: *** [main.o] Error 1
>
>
> Any idea ?
>
> Thx
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Tue Mar  2 22:01:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 3 Mar 2010 03:01:46 -0300
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
	<93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>
Message-ID: <ebbc2d551003022201p6a775347wbeea66c8d0e4c32e@mail.gmail.com>

I have added you on GTalk, let me know if you need help.

Regards,
Jo?o Mesquita


On Sat, Feb 20, 2010 at 11:12 PM, ??? <spiritonly at gmail.com> wrote:

> Oh, My God! You are the developer of mod_khomp! I read your blog, checked
> out mod_khomp from google code, and found it had not update for a long time.
> So what is your roadmap about mod_khomp?
>
> I share my snippet in FS pastebin, you can find it with '
> http://pastebin.freeswitch.org/12192 '.
>
> My MSN is spiritonly at live.cn.
> My Gtalk is spiritonly at gmail.com.
> Hope we can keep connected.
>
>
> 2010/2/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>  I developed the current implementation of mod_khomp. I wouldn't take it as
>> an example for anything since there has been no activity there for the past
>> 4 months. If you care to share a snippet of your code, maybe we can help
>> better.
>>
>>
>> JM
>>
>>
>>
>> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
>>
>>> Do you know mod_khomp? You can found it in FS wiki. I am developing an
>>> endpoint module like it.
>>> So you can give me some advice to bridge two session?
>>>
>>>
>>> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> But the bigger question is what protocol are you doing that you have to
>>>> create your own endpoint module?
>>>>
>>>> /b
>>>>
>>>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>>>
>>>> > You should look at read_frame and write_frame implementations of other
>>>> endpoint modules.
>>>> >
>>>> > This should pretty much tell you how things work...
>>>> >
>>>> > Jo?o Mesquita
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar  2 22:26:54 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 01:26:54 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705><COL117-W5563292C886257AFC56548B9420@phx.gbl><D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705><BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com><D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
	<ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>
Message-ID: <13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>

Hi Joao,

thanks for your answer.
I have a standard RPM QT last update for Fedora10 64bits.
so I really don't know what path FScomm needs

Regards

Franck
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 1:00 AM
  Subject: Re: [Freeswitch-users] FScomm


  This is definitely a problem with your Qt installation. You need to be able to find the libs to link it, otherwise, unresolved symbols everywhere!


  Regards,
  Jo?o Mesquita
  FSComm Developer



  On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:


    ----- Original Message -----
    From: "Michael Jerris" <mike at jerris.com>

    To: <freeswitch-users at lists.freeswitch.org>

    Sent: Wednesday, February 24, 2010 3:40 AM

    Subject: Re: [Freeswitch-users] FScomm



    On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

    >
    > ----- Original Message -----
    > From: Jeff Lenk
    > To: freeswitch-users at lists.freeswitch.org
    > Sent: Tuesday, February 23, 2010 5:05 PM
    > Subject: Re: [Freeswitch-users] FScomm
    >
    >
    > http://wiki.freeswitch.org/wiki/FSComm#Linux
    >
    > you must run those from the FSComm directory
    >

    ?.

    >
    > It's what I did,
    > but from FS trunk, inside fscomm directory,
    > there s only
    >
    > account.cpp  conf                fshost.h        mainwindow.ui
    > resources.qrc
    > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
    > call.cpp     FSComm.pro          mainwindow.cpp  preferences
    > call.h       fshost.cpp          mainwindow.h    resources
    >


    Read those installation instructions again and do them step by step, you
    skipped one.
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



    Ok now I have

    [root at node250 fscomm]# qmake
    WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
    in SOURCES
    WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
    HEADERS
    WARNING: Found potential symbol conflict of prefdialog.cpp
    (preferences/prefdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of prefdialog.h
    (preferences/prefdialog.h) in HEADERS
    WARNING: Found potential symbol conflict of accountdialog.cpp
    (preferences/accountdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of accountdialog.h
    (preferences/accountdialog.h) in HEADERS

    [root at node250 fscomm]# qmake
    WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
    in SOURCES
    WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
    HEADERS
    WARNING: Found potential symbol conflict of prefdialog.cpp
    (preferences/prefdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of prefdialog.h
    (preferences/prefdialog.h) in HEADERS
    WARNING: Found potential symbol conflict of accountdialog.cpp
    (preferences/accountdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of accountdialog.h
    (preferences/accountdialog.h) in HEADERS
    [root at node250 fscomm]# make
    Makefile:278: warning: overriding commands for target `prefdialog.o'
    Makefile:215: warning: ignoring old commands for target `prefdialog.o'
    Makefile:285: warning: overriding commands for target `accountdialog.o'
    Makefile:234: warning: ignoring old commands for target `accountdialog.o'
    Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
    Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
    Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
    Makefile:307: warning: ignoring old commands for target
    `moc_accountdialog.o'
    Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
    Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
    Makefile:350: warning: overriding commands for target
    `preferences/moc_prefdialog.cpp'
    Makefile:332: warning: ignoring old commands for target
    `preferences/moc_prefdialog.cpp'
    Makefile:353: warning: overriding commands for target
    `preferences/moc_accountdialog.cpp'
    Makefile:341: warning: ignoring old commands for target
    `preferences/moc_accountdialog.cpp'
    g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
     -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG
     -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default
     -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
     -o main.o main.cpp
    main.cpp:31:25: error: QSplashScreen: No such file or directory
    In file included from main.cpp:32:
    mainwindow.h:34:23: error: QMainWindow: No such file or directory
    mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
    mainwindow.h:36:25: error: QSignalMapper: No such file or directory
    mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
    In file included from mainwindow.h:39,
                    from main.cpp:32:
    ./fshost.h:32:19: error: QThread: No such file or directory
    ./fshost.h:33:17: error: QHash: No such file or directory
    ./fshost.h:34:26: error: QSharedPointer: No such file or directory
    In file included from ./fshost.h:36,
                    from mainwindow.h:39,
                    from main.cpp:32:
    ./call.h:32:18: error: QtCore: No such file or directory
    ./call.h:33:19: error: QString: No such file or directory
    In file included from mainwindow.h:42,
                    from main.cpp:32:
    preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
    preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
    directory
    preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
    In file included from ./fshost.h:37,
                    from mainwindow.h:39,
                    from main.cpp:32:
    ./account.h:18: error: expected constructor, destructor, or type conversion
    before ?static?
    In file included from mainwindow.h:39,
                    from main.cpp:32:
    ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
    /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
    QThread?
    ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:46: error: expected ?;? before ?<? token
    ./fshost.h:47: error: expected `;' before ?QSharedPointer?
    ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:47: error: expected ?;? before ?<? token
    ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:48: error: template argument 1 is invalid
    ./fshost.h:48: error: expected unqualified-id before ?>? token
    ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:49: error: expected ?;? before ?<? token
    ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:50: error: expected ?;? before ?<? token
    ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:51: error: expected ?;? before ?<? token
    ./fshost.h:52: error: ?QSharedPointer? has not been declared
    ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:60: error: ?QSharedPointer? has not been declared
    ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:61: error: ?QSharedPointer? has not been declared
    ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:62: error: ?QSharedPointer? has not been declared
    ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:63: error: ?QSharedPointer? has not been declared
    ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:64: error: ?QSharedPointer? has not been declared
    ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:65: error: ?QSharedPointer? has not been declared
    ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:66: error: ?QSharedPointer? has not been declared
    ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:67: error: ?QSharedPointer? has not been declared
    ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:71: error: ?QSharedPointer? has not been declared
    ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:78: error: template argument 2 is invalid
    ./fshost.h:78: error: expected unqualified-id before ?>? token
    ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:79: error: template argument 2 is invalid
    ./fshost.h:79: error: expected unqualified-id before ?>? token
    ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
    In file included from mainwindow.h:42,
                    from main.cpp:32:
    preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
    QDialog?
    /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
    QDialog?
    preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
    ?QSettings? with no type
    preferences/prefdialog.h:31: error: expected ?;? before ?*? token
    In file included from main.cpp:32:
    mainwindow.h:48: error: expected class-name before ?{? token
    mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
    mainwindow.h:71: error: ?QSharedPointer? has not been declared
    mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:72: error: ?QSharedPointer? has not been declared
    mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:73: error: ?QSharedPointer? has not been declared
    mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:74: error: ?QSharedPointer? has not been declared
    mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:75: error: ?QSharedPointer? has not been declared
    mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:78: error: ?QSharedPointer? has not been declared
    mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:79: error: ?QSharedPointer? has not been declared
    mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:80: error: ?QSharedPointer? has not been declared
    mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
    mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
    mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
    no type
    mainwindow.h:86: error: expected ?;? before ?*? token
    mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
    with no type
    mainwindow.h:88: error: expected ?;? before ?*? token
    main.cpp: In function ?int main(int, char**)?:
    main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
    type
    main.cpp:42: error: ?QSplashScreen? was not declared in this scope
    main.cpp:42: error: ?splash? was not declared in this scope
    main.cpp:42: error: expected type-specifier before ?QSplashScreen?
    main.cpp:42: error: expected `;' before ?QSplashScreen?
    main.cpp:48: error: no matching function for call to
    ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
    /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
    QObject::connect(const QObject*, const char*, const QObject*, const char*,
    Qt::ConnectionType)
    /usr/include/QtCore/qobject.h:307: note:                 bool
    QObject::connect(const QObject*, const char*, const char*,
    Qt::ConnectionType) const
    main.cpp:49: error: ?class FSHost? has no member named ?start?
    main.cpp:41: warning: unused variable ?image?
    ./fshost.h: At global scope:
    ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined
    but not used
    make: *** [main.o] Error 1


    Any idea ?

    Thx




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





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


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From dome at tel.co.th  Tue Mar  2 22:38:58 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 3 Mar 2010 13:38:58 +0700
Subject: [Freeswitch-users] Nibblebill for NoSQL
Message-ID: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>

Dear All,
     I use nibllebill + postgresql for longtime. it' work wel when
concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
many problem and slow. so i have 2 choice.
1. upgrade hardware , add more memory may be 32 GB and change HDD to
SSD (or IOdrive ).
2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
I choose 2. and use tokyo tyrant. because it's persistant and support
memcache API.
  Today i create new module base on nibblebill but use memcacahe API
connect to tokyo tyrant. it's look fast. and work without any problem.
  If someone want to use my module I'l acttach to JIRA and create WIKI page.

Please let's me know if you want.

BG
Dome C.


From infos at madovsky.org  Tue Mar  2 22:49:34 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 01:49:34 -0500
Subject: [Freeswitch-users] Nibblebill for NoSQL
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <1B6F4F74BC724CCC8FB842F63D0279E0@MOBILEE1705>

Hi Dome,

postgresql is a very powerful DB that can accepts a lot of simultaneous 
connections
it needs to be set carefully to fit your memory and HD.
try google postgresql tuning
However the other solution can be interesting too.

Regards

F

----- Original Message ----- 
From: "Dome Charoenyost" <dome at tel.co.th>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 03, 2010 1:38 AM
Subject: [Freeswitch-users] Nibblebill for NoSQL


> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI 
> page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From mayamatakeshi at gmail.com  Wed Mar  3 00:06:28 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 3 Mar 2010 17:06:28 +0900
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
Message-ID: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>

Hello,
was any timing related change done recently in FS?
I have updated to trunk r. 16873 (was just some 200 revisions behind) and
now I'm seeing several messages like this when I try as little as 5 calls
per second:
2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle Error!
111

When this happens, FS refuses the related call with "503 Maximum Calls In
Progress" although, I'm using default config files (sessions-per-second=30).
So it seems to me FS is wrongly aggregating incoming calls on the same
interval.

I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
FS never showed this before in any of them.

regards,
takeshi
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From jonas.gauffin at gmail.com  Wed Mar  3 00:07:09 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 3 Mar 2010 09:07:09 +0100
Subject: [Freeswitch-users] OT: SIP and IP spoofing
Message-ID: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>

Hello,

My sip gateway provider are using both IP address locking (only my servers
IP addresses can use my gateway account) and Digest authentication on every
call.
I asked why and they said that the account would be vulnerable to IP
spoofing otherwise. Is that possible? I mean, if someone fakes my servers IP
address in the packets, shouldn't the responses be sent back to my server
and not the one creating the fake packets? Are there any other reasons to
use both ip locking and digest authentication?

Regards,
  Jonas
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From red.rain.seven at gmail.com  Wed Mar  3 00:07:42 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 3 Mar 2010 16:07:42 +0800
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <59ad9ca11003030007m262d1544hc72af6d2175c072e@mail.gmail.com>

Dome

I am very interested in your module. Please add it to JIRA and create a WIKI
page for it. And does everything else that used to be on your postgresql now
being migrated to the memcache API + tokyo tyrant?



On Wed, Mar 3, 2010 at 2:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jaybinks at gmail.com  Wed Mar  3 00:42:18 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 3 Mar 2010 18:42:18 +1000
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>

im VERY interested in this...
( and the use of NoSQL in general with FS )

please can you email over what you have, or post it to jira.

J

On Wed, Mar 3, 2010 at 4:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From viper at fx-services.com  Wed Mar  3 05:49:54 2010
From: viper at fx-services.com (Robin Vleij)
Date: Wed, 03 Mar 2010 14:49:54 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
Message-ID: <4B8E6902.90106@fx-services.com>

On 2010-03-02 16:02, Jonas Gauffin wrote:

Hi Jonas,

> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes
> incorrect (I got the trace from my sip provider).

Your problem is not a bug. The problem is that your SIP provider has two
proxies (that you probably find via SRV). Instead of using one gateway
profile that selects either of them, you have to configure both gateways
as separate elements. That way FS will use the nonce it gets in the 407,
instead of mixing them up between the two proxies (who don't have state).

You either use one gateway for outbound and the other one as failover in
the bridge app, or you use mod_distribute to distribute.

/Robin


From phunk0000 at hotmail.com  Wed Mar  3 06:19:37 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 3 Mar 2010 09:19:37 -0500
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
Message-ID: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>

He group confirm feature seems like it should work, but I can't figure out
where in the default dialplan to put this line so that the audio file is
played for everycall made to a local extension. here us the
group_confirm_playback action, I am wondering where it should go.

 

<action application="bridge" data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

 

Here is the local extension portion of the default dialplan where I think it
should go, but I can't figure out exactly where.  Thanks!

 

<extension name="Local_Extension">

                  <condition field="destination_number"
expression="(^\d{3,6}$)">

                    <!-- <action application="javascript"
data="pbxpalintro.js" /> -->

                   

                    <action application="set" data="dialed_extension=$1"/>

                    <action application="export"
data="dialed_extension=$1"/>

                    

                    <action application="bridge"
data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

                   

                    <!-- bind_meta_app can have these args <key> [a|b|ab]
[a|b|o|s] <app> -->

                    <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>

                    <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

                    <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>

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

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

                    <action application="set" data="call_timeout=30"/>

                    <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->

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

                    <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,
NO_ROUTE_DESTINATION"/> -->

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

                    <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>

                    <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>

                    <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>

                    <!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name
} var sip_secure_media)}"/>-->

                    <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

                    <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

                    <action application="answer"/>

                    <action application="sleep" data="1000"/>

                    <action application="voicemail" data="default
${domain_name} ${dialed_extension}"/>

                  </condition>

                </extension>

 

 

 

 

 

 

 

                   

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, March 02, 2010 11:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

 

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks

Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC 

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From brian at freeswitch.org  Wed Mar  3 06:30:35 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 08:30:35 -0600
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
	<BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
Message-ID: <28735474-6C43-4B1C-A815-CEB9CF39128E@freeswitch.org>

you need user/user at domain not user/user%domain,,,, The faq outlines this syntax used for sofia/profiel/user%domain notation.  Please don't confuse the two.

/b

On Mar 3, 2010, at 8:19 AM, Todd wrote:

> <action application="bridge" data="{group_confirm_file=playback /usr/local/freeswitch/sounds/en/pbxpalintro.wav, group_confirm_key=exec,call_timeout=60} user/${dialed_extension}%${domain_name}"/>

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From phunk0000 at hotmail.com  Wed Mar  3 06:37:33 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 3 Mar 2010 09:37:33 -0500
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
	<BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
Message-ID: <BLU110-DS946BAEF42517F230E5CB4A73A0@phx.gbl>

Or maybe there is an easier way to play an audio file to whoever answers the
phone before the call is bridged..  And is it best practice to put these
changes in the default dial plan or somewhere else?  Still kinda new to
this, Thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd
Sent: Wednesday, March 03, 2010 9:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

He group confirm feature seems like it should work, but I can't figure out
where in the default dialplan to put this line so that the audio file is
played for everycall made to a local extension. here us the
group_confirm_playback action, I am wondering where it should go.

 

<action application="bridge" data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

 

Here is the local extension portion of the default dialplan where I think it
should go, but I can't figure out exactly where.  Thanks!

 

<extension name="Local_Extension">

                  <condition field="destination_number"
expression="(^\d{3,6}$)">

                    <!-- <action application="javascript"
data="pbxpalintro.js" /> -->

                   

                    <action application="set" data="dialed_extension=$1"/>

                    <action application="export"
data="dialed_extension=$1"/>

                    

                    <action application="bridge"
data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

                   

                    <!-- bind_meta_app can have these args <key> [a|b|ab]
[a|b|o|s] <app> -->

                    <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>

                    <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

                    <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>

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

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

                    <action application="set" data="call_timeout=30"/>

                    <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->

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

                    <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,
NO_ROUTE_DESTINATION"/> -->

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

                    <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>

                    <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>

                    <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>

                    <!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name
} var sip_secure_media)}"/>-->

                    <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

                    <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

                    <action application="answer"/>

                    <action application="sleep" data="1000"/>

                    <action application="voicemail" data="default
${domain_name} ${dialed_extension}"/>

                  </condition>

                </extension>

 

 

 

 

 

 

 

                   

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, March 02, 2010 11:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

 

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks

Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC 

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From brent at overthewire.com.au  Tue Mar  2 22:19:35 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Wed, 3 Mar 2010 16:19:35 +1000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com> 
	<3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
Message-ID: <ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>

I suspect you would be better off with some virtualization platform more
like OpenVZ ('containers' rather than VM's) - and probably without any
hardware
installed (ie SIP to SIP - leave the gateways to a different part of the
network).

We've run some Asterisk instances in Proxmox (
http://pve.proxmox.com/wiki/Main_Page) and while it's early days, it looks
pretty
promising.

Brent

On Wed, Mar 3, 2010 at 3:27 PM, Brian May <brian at microcomaustralia.com.au>wrote:

> On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> > Is anyone running FreeSWITCH virtualized (xen or vmware) in
> > production?  I saw ec2 and xen mentioned a few places.  If so, are
> > there tips or tricks for handle timer issues which affect virtual
> > machines in general, such as recommended divider or clocksource
> > settings?  I'm seeing funny timing issues while testing with audio
> > recording and playback that speed up or slow down.
>
> I tried tdm400+xen DOM0+asterisk years ago, on a single processor
> machine but gave up. zttest returned good results, but analogue
> quality has intermittently very poor quality and it made telephone
> conversations difficult to understand. Especially a problem when
> hourly cron jobs would start on all VMs at exactly the same time, but
> was a problem at other times too. Plus the fact people who used the
> phone more often then me didn't report the problem, so I thought it
> was fine.
>
> Plus the general fact I didn't really like the idea of running
> Asterisk on DOM0, and running it on DOMU seemed to imply even more
> overheard, as DOM0 would have to process the interrupts and pass them
> to the DOMU.
>
> Unfortunately I don't remember what scheduler I used, maybe sedf?
>
> I suspect a multi-processor system would be a lot better suited for
> the task. Also might be OK if you don't use zaptel hardware... I seem
> to remember these cards generate a huge number of interrupts.
>
> While this was with Asterisk, I suspect my results were not specific
> to Asterisk in anyway.
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From sos at sokhapkin.dyndns.org  Wed Mar  3 07:17:00 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 3 Mar 2010 10:17:00 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
	<ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>
Message-ID: <201003031017.00850.sos@sokhapkin.dyndns.org>

I second that, do not use VMs. I run tens of asterisks under linux vserver on 
the single box, no issues. I do not expect FS will behave differently under 
vserver.

On Wednesday 03 March 2010, Brent Paddon wrote:
> I suspect you would be better off with some virtualization platform more
> like OpenVZ ('containers' rather than VM's) - and probably without any
> hardware
> installed (ie SIP to SIP - leave the gateways to a different part of the
> network).
>
> We've run some Asterisk instances in Proxmox (
> http://pve.proxmox.com/wiki/Main_Page) and while it's early days, it looks
> pretty
> promising.
>
> Brent
>
> On Wed, Mar 3, 2010 at 3:27 PM, Brian May 
<brian at microcomaustralia.com.au>wrote:
> > On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> > > Is anyone running FreeSWITCH virtualized (xen or vmware) in
> > > production?  I saw ec2 and xen mentioned a few places.  If so, are
> > > there tips or tricks for handle timer issues which affect virtual
> > > machines in general, such as recommended divider or clocksource
> > > settings?  I'm seeing funny timing issues while testing with audio
> > > recording and playback that speed up or slow down.
> >
> > I tried tdm400+xen DOM0+asterisk years ago, on a single processor
> > machine but gave up. zttest returned good results, but analogue
> > quality has intermittently very poor quality and it made telephone
> > conversations difficult to understand. Especially a problem when
> > hourly cron jobs would start on all VMs at exactly the same time, but
> > was a problem at other times too. Plus the fact people who used the
> > phone more often then me didn't report the problem, so I thought it
> > was fine.
> >
> > Plus the general fact I didn't really like the idea of running
> > Asterisk on DOM0, and running it on DOMU seemed to imply even more
> > overheard, as DOM0 would have to process the interrupts and pass them
> > to the DOMU.
> >
> > Unfortunately I don't remember what scheduler I used, maybe sedf?
> >
> > I suspect a multi-processor system would be a lot better suited for
> > the task. Also might be OK if you don't use zaptel hardware... I seem
> > to remember these cards generate a huge number of interrupts.
> >
> > While this was with Asterisk, I suspect my results were not specific
> > to Asterisk in anyway.
> > --
> > Brian May <brian at microcomaustralia.com.au>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org




From javieraristizabal at gmail.com  Wed Mar  3 07:19:26 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Wed, 3 Mar 2010 10:19:26 -0500
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>
Message-ID: <b6f11f301003030719gb64ec72k5f38c031f1ce9049@mail.gmail.com>

Hi.. I'm interesting too..

Thanks :)

On Wed, Mar 3, 2010 at 3:42 AM, jay binks <jaybinks at gmail.com> wrote:

> im VERY interested in this...
> ( and the use of NoSQL in general with FS )
>
> please can you email over what you have, or post it to jira.
>
> J
>
> On Wed, Mar 3, 2010 at 4:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:
>
>> Dear All,
>>     I use nibllebill + postgresql for longtime. it' work wel when
>> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
>> many problem and slow. so i have 2 choice.
>> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
>> SSD (or IOdrive ).
>> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
>> I choose 2. and use tokyo tyrant. because it's persistant and support
>> memcache API.
>>  Today i create new module base on nibblebill but use memcacahe API
>> connect to tokyo tyrant. it's look fast. and work without any problem.
>>  If someone want to use my module I'l acttach to JIRA and create WIKI
>> page.
>>
>> Please let's me know if you want.
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar  3 08:19:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Mar 2010 08:19:11 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003030819ld02171ub6bc4cbc191eeb94@mail.gmail.com>

Hello all. We have a nice agenda today:

http://bit.ly/9zDiBq

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

Talk to you all soon,
Michael
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From anthony.minessale at gmail.com  Wed Mar  3 08:23:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Mar 2010 10:23:47 -0600
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
In-Reply-To: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
References: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
Message-ID: <191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>

try again 16880 or higher

On Wed, Mar 3, 2010 at 2:06 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> was any timing related change done recently in FS?
> I have updated to trunk r. 16873 (was just some 200 revisions behind) and
> now I'm seeing several messages like this when I try as little as 5 calls
> per second:
> 2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle
> Error! 111
>
> When this happens, FS refuses the related call with "503 Maximum Calls In
> Progress" although, I'm using default config files (sessions-per-second=30).
> So it seems to me FS is wrongly aggregating incoming calls on the same
> interval.
>
> I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
> FS never showed this before in any of them.
>
> regards,
> takeshi
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From chris at fowler.cc  Wed Mar  3 09:17:02 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 3 Mar 2010 12:17:02 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>

Hi Rob,

We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily supporting 100 Polycom users, conference bridges etc.  Been running for over a year.  We also use FlowRoute as our PSTN->SIP Interface, and also Skype Business SIP.  Our users are scattered across the globe - so having the box sitting on AWS' network infrastructure is key to avoiding issues with latency, jitter, packet loss (i.e. I don't think we could afford the connectivity AWS gives us if we had to provision this in-house).

There were no special tricks; you do need to modify/override the following with the box's Elastic IP (EIP).

modify conf/vars.xml and update
  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>

  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=flowroute.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>

conf/sip_profiles/internal.xml
 <param name="aggressive-nat-detection" value="true"/>
 <param name="multiple-registrations" value="true"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="NDLB-received-in-nat-reg-contact" value="true"/>
 <param name="NDLB-force-rport" value="true"/>
 <param name="NDLB-broken-auth-hash" value="true"/>
 <param name="enable-timer" value="false"/>

conf/sip_profiles/external.xml
 <param name="aggressive-nat-detection" value="true"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="NDLB-force-rport" value="true"/>

conf/autoload/switch.conf.xml
    <param name="rtp-start-port" value="16384"/>
    <param name="rtp-end-port" value="32768"/>


Cheers, Chris.
--
RightScale, Inc.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Tuesday, March 02, 2010 9:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Virtualized FreeSWITCH

Hi all,

Is anyone running FreeSWITCH virtualized (xen or vmware) in
production?  I saw ec2 and xen mentioned a few places.  If so, are
there tips or tricks for handle timer issues which affect virtual
machines in general, such as recommended divider or clocksource
settings?  I'm seeing funny timing issues while testing with audio
recording and playback that speed up or slow down.

I have tried both paravirt and hvm.  The default xen kernel has a
timer of 250, so I tried to hvm instead using the stock 5.3 kernel
with the 1khz timer but didn't see much improvement.

All of this was tested on 64-bit CentOS 5.3.
- Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=1000


I'm sure physical hosts are the de-facto best practice, but are
virtual machines not worth it at all?

Cheers,
Rob

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


From mayamatakeshi at gmail.com  Wed Mar  3 09:22:17 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 4 Mar 2010 02:22:17 +0900
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
In-Reply-To: <191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>
References: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
	<191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>
Message-ID: <15b9404e1003030922p716b1c0cj3999b1d4ff5cf2ad@mail.gmail.com>

It is OK now, Anthony.
Thanks.


On Thu, Mar 4, 2010 at 1:23 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try again 16880 or higher
>
> On Wed, Mar 3, 2010 at 2:06 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> Hello,
>> was any timing related change done recently in FS?
>> I have updated to trunk r. 16873 (was just some 200 revisions behind) and
>> now I'm seeing several messages like this when I try as little as 5 calls
>> per second:
>> 2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle
>> Error! 111
>>
>> When this happens, FS refuses the related call with "503 Maximum Calls In
>> Progress" although, I'm using default config files (sessions-per-second=30).
>> So it seems to me FS is wrongly aggregating incoming calls on the same
>> interval.
>>
>> I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
>> FS never showed this before in any of them.
>>
>> regards,
>> takeshi
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar  3 09:23:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Mar 2010 09:23:39 -0800
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <87f2f3b91003030923v24deedc1wd034f1c1337c3536@mail.gmail.com>

Chris,

This is good stuff! Would you be willing to compare this to what's on the
wiki and update the wiki as needed? I believe we have some EC2 information
on the wiki...

Thanks,
MC

On Wed, Mar 3, 2010 at 9:17 AM, Chris Fowler <chris at fowler.cc> wrote:

> Hi Rob,
>
> We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily supporting
> 100 Polycom users, conference bridges etc.  Been running for over a year.
>  We also use FlowRoute as our PSTN->SIP Interface, and also Skype Business
> SIP.  Our users are scattered across the globe - so having the box sitting
> on AWS' network infrastructure is key to avoiding issues with latency,
> jitter, packet loss (i.e. I don't think we could afford the connectivity AWS
> gives us if we had to provision this in-house).
>
> There were no special tricks; you do need to modify/override the following
> with the box's Elastic IP (EIP).
>
> modify conf/vars.xml and update
>  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>
>
>  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=flowroute.com
> "/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>
>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>
>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>
>
> conf/sip_profiles/internal.xml
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="multiple-registrations" value="true"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>  <param name="NDLB-force-rport" value="true"/>
>  <param name="NDLB-broken-auth-hash" value="true"/>
>  <param name="enable-timer" value="false"/>
>
> conf/sip_profiles/external.xml
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="NDLB-force-rport" value="true"/>
>
> conf/autoload/switch.conf.xml
>    <param name="rtp-start-port" value="16384"/>
>    <param name="rtp-end-port" value="32768"/>
>
>
> Cheers, Chris.
> --
> RightScale, Inc.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
> Sent: Tuesday, March 02, 2010 9:05 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Virtualized FreeSWITCH
>
> Hi all,
>
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production?  I saw ec2 and xen mentioned a few places.  If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings?  I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.
>
> I have tried both paravirt and hvm.  The default xen kernel has a
> timer of 250, so I tried to hvm instead using the stock 5.3 kernel
> with the 1khz timer but didn't see much improvement.
>
> All of this was tested on 64-bit CentOS 5.3.
> - Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
> SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
> Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
> 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=1000
>
>
> I'm sure physical hosts are the de-facto best practice, but are
> virtual machines not worth it at all?
>
> Cheers,
> Rob
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rob4manhere at gmail.com  Wed Mar  3 09:47:35 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 3 Mar 2010 11:47:35 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <3F2BADF0-DE12-42A8-9F39-2B89FCCB45DD@gmail.com>

Hi Chris,

Thanks for sharing what you're doing.  That's very helpful.  Which AMI  
did you deploy from?  I'm wondering what your ec2 kernel timer is set  
to.

Thanks!
Rob

On Mar 3, 2010, at 11:17 AM, Chris Fowler wrote:

> Hi Rob,
>
> We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily  
> supporting 100 Polycom users, conference bridges etc.  Been running  
> for over a year.  We also use FlowRoute as our PSTN->SIP Interface,  
> and also Skype Business SIP.  Our users are scattered across the  
> globe - so having the box sitting on AWS' network infrastructure is  
> key to avoiding issues with latency, jitter, packet loss (i.e. I  
> don't think we could afford the connectivity AWS gives us if we had  
> to provision this in-house).
>
> There were no special tricks; you do need to modify/override the  
> following with the box's Elastic IP (EIP).
>
> modify conf/vars.xml and update
>  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>
>
>  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
>  <X-PRE-PROCESS cmd="set"  
> data="default_provider_from_domain=flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>
>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>
>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>
>
> conf/sip_profiles/internal.xml
> <param name="aggressive-nat-detection" value="true"/>
> <param name="multiple-registrations" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> <param name="NDLB-received-in-nat-reg-contact" value="true"/>
> <param name="NDLB-force-rport" value="true"/>
> <param name="NDLB-broken-auth-hash" value="true"/>
> <param name="enable-timer" value="false"/>
>
> conf/sip_profiles/external.xml
> <param name="aggressive-nat-detection" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> <param name="NDLB-force-rport" value="true"/>
>
> conf/autoload/switch.conf.xml
>    <param name="rtp-start-port" value="16384"/>
>    <param name="rtp-end-port" value="32768"/>
>
>
> Cheers, Chris.
> --
> RightScale, Inc.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman
> Sent: Tuesday, March 02, 2010 9:05 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Virtualized FreeSWITCH
>
> Hi all,
>
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production?  I saw ec2 and xen mentioned a few places.  If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings?  I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.
>
> I have tried both paravirt and hvm.  The default xen kernel has a
> timer of 250, so I tried to hvm instead using the stock 5.3 kernel
> with the 1khz timer but didn't see much improvement.
>
> All of this was tested on 64-bit CentOS 5.3.
> - Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
> SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
> Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
> 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=1000
>
>
> I'm sure physical hosts are the de-facto best practice, but are
> virtual machines not worth it at all?
>
> Cheers,
> Rob
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From robert.hadley at teotech.com  Wed Mar  3 09:52:15 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 3 Mar 2010 09:52:15 -0800
Subject: [Freeswitch-users] Delete custom voicemail greetings
In-Reply-To: <7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>
	<7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
Message-ID: <8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>

Thanks for replying.  I tried make install samples but it didn't work.

 

My original question should have been "How would an individual user delete a
custom voicemail greeting and reset back to the default greeting?"

 

Thanks,

Robert

 

  _____  

From: Madovsky [mailto:infos at madovsky.org] 
Sent: Tuesday, March 02, 2010 6:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

 

make install samples

----- Original Message ----- 

From: Robert <mailto:robert.hadley at teotech.com>  Hadley 

To: freeswitch-users at lists.freeswitch.org 

Sent: Tuesday, March 02, 2010 4:31 PM

Subject: [Freeswitch-users] Delete custom voicemail greetings

 

Hi FS users,

 

How do I delete custom voicemail greetings and go back to the default
greeting? 

 

Thanks,

Robert


  _____  


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

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From infos at madovsky.org  Wed Mar  3 10:03:26 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 13:03:26 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com><7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
	<8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>
Message-ID: <380273FE029E45BCB5D83B505372F27A@MOBILEE1705>

try to loo at the FS sources folder, I'm sure there will be the right .wav for that
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 12:52 PM
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings


  Thanks for replying.  I tried make install samples but it didn't work.

   

  My original question should have been "How would an individual user delete a custom voicemail greeting and reset back to the default greeting?"

   

  Thanks,

  Robert

   


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

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Tuesday, March 02, 2010 6:47 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

   

  make install samples

    ----- Original Message ----- 

    From: Robert Hadley 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Tuesday, March 02, 2010 4:31 PM

    Subject: [Freeswitch-users] Delete custom voicemail greetings

     

    Hi FS users,

     

    How do I delete custom voicemail greetings and go back to the default greeting? 

     

    Thanks,

    Robert


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

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



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch at cartissolutions.com  Wed Mar  3 10:04:26 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Wed, 03 Mar 2010 12:04:26 -0600
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <4B8EA4AA.4010602@cartissolutions.com>


On 03/03/2010 12:38 AM, Dome Charoenyost wrote:

<snip snip>

>    Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>    If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
>    

Maybe it would be even more useful to provide general functionality that 
can be shared amongst multiple components of freeswitch.  That would 
make it all the more useful.  However, seeing that I'm not a big fan of 
the NoSQL data engines, I don't know if this is possible.  In my 
opinion, most NoSQL is just key=>value pairs, and is basically a 
reinvention of the wheel that RDBMS's had taken care of years ago...   
Not meaning to open up a discussion about the virtues of NoSQL (and 
certainly not a flamewar)...


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com




From helmut.kuper at ewetel.de  Wed Mar  3 10:16:23 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 03 Mar 2010 19:16:23 +0100
Subject: [Freeswitch-users] RFC3326 Reason Header field
Message-ID: <4B8EA777.9080207@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I'm looking for a way to add a reason header field (like 'Reason:
SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
send by bridge application when a call times out or was denied.

I would like to set it on demand in dialplan, e.g. when a queue is
calling a queue member and I don't want to have that call in the phone's
missed calls list.

Is there a way in FS to do this, yet?

best regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
hcWP9lUEFxINGpyLXgfDKLU=
=8qjy
-----END PGP SIGNATURE-----


From Russell.Mosemann at cune.org  Wed Mar  3 10:19:09 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 3 Mar 2010 18:19:09 -0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>

Chris Fowler <chris at fowler.cc> said:

> There were no special tricks; you do need to modify/override the
> following with the box's Elastic IP (EIP).
 
This would be helpful to have in the wiki along with any other tips for
your virtual environment. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Wed Mar  3 10:21:49 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 12:21:49 -0600
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <4B8EA777.9080207@ewetel.de>
References: <4B8EA777.9080207@ewetel.de>
Message-ID: <0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>

FreeSWITCH already does this for you on cancel when the call is answered elsewhere.

/b

On Mar 3, 2010, at 12:16 PM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hello,
> 
> I'm looking for a way to add a reason header field (like 'Reason:
> SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
> send by bridge application when a call times out or was denied.
> 
> I would like to set it on demand in dialplan, e.g. when a queue is
> calling a queue member and I don't want to have that call in the phone's
> missed calls list.
> 
> Is there a way in FS to do this, yet?
> 
> best regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> 
> iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
> hcWP9lUEFxINGpyLXgfDKLU=
> =8qjy
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rupa at rupa.com  Wed Mar  3 10:25:37 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 3 Mar 2010 12:25:37 -0600
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <4B8EA4AA.4010602@cartissolutions.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
Message-ID: <b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>

On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

>
>
> Maybe it would be even more useful to provide general functionality that
> can be shared amongst multiple components of freeswitch.  That would
> make it all the more useful.  However, seeing that I'm not a big fan of
> the NoSQL data engines, I don't know if this is possible.  In my
> opinion, most NoSQL is just key=>value pairs, and is basically a
> reinvention of the wheel that RDBMS's had taken care of years ago...
> Not meaning to open up a discussion about the virtues of NoSQL (and
> certainly not a flamewar)...
>
>
I created a mod_memcache already and I intend to do a mod_redis at some
point.  I'm not a huge fan of NoSQL but there are definite uses for it.

General functionality:  I could be convinced to come up with a general api
for distributed key/value store with mod_memcache and mod_redis providing
implementation.  The problem is that the NoSQL stuff isn't very consistent
so even though both support key/value their behavior can significantly
differ.  eg: updating a value in memcached doesn't bump the expire time but
does in redis.  Also, redis has support for a much more robust set of value
types and operators.

Anyway, a generic distributed key/value api might look like:

dhash <backend> set key value [expire]
dhash <backend> setnx key value [expire]  # only set if it doesn't already
exist
dhash <backend> get key [...] # support multiple keys
dhash <backend> <increment|decrement> key [step]
dhash <backend> del key

where backend would be memcache or redis or some other implementation.

Notice I didn't even touch things like hashing to support sharding,
failover, etc.


>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
-- 
-Rupa
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From jason at jasonjgw.net  Wed Mar  3 10:42:34 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 03 Mar 2010 12:42:34 -0600
Subject: Tienes una invitación de Jason White a ClubWNC
Message-ID: <E1NmtWk-0004pD-Oi@eclipse.websitewelcome.com>

ClubWNC&nbsp;freeswitch-users: Tienes una invitaci?n de Jason White a ClubWNC 	&nbsp; 	Hola  
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2465                           	Jason White ha invitado a freeswitch-users a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From stevendt at primrosebank.net  Wed Mar  3 11:39:49 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 19:39:49 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payload
	check.
Message-ID: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>

Hi,

can someone please advise what the above error message means ?

"[ERR] switch_rtp.c:2196 Failed DTMF payload check."

It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?

regards
Dave

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From brian at freeswitch.org  Wed Mar  3 11:50:42 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 13:50:42 -0600
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payload
	check.
In-Reply-To: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
Message-ID: <59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>

It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.

/b

On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:

>  
> It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?
>  

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From stevendt at primrosebank.net  Wed Mar  3 12:06:53 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:06:53 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF
	payloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
	<59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>
Message-ID: <E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>

Hi,

thanks Brian.

The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

regards
Dave
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 7:50 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


  It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


  /b


  On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



    It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?





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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mrene_lists at avgs.ca  Wed Mar  3 12:09:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 15:09:31 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF
	payloadcheck.
In-Reply-To: <E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
	<59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>
	<E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
Message-ID: <3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>

I don't recall any rfc2833 changes in the past 2 weeks. I would like  
to see a packet capture of those bogus rtp packets though.

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




On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:

> Hi,
>
> thanks Brian.
>
> The device is a Thomson ST2030 which has a basic listing on the  
> Interop Wiki page, but does not specifically mention rfc2833  
> (although rtp is marked as "not tested"). I have not seen this  
> message before while running the FS version that I'm currently using  
> 16543 - yes, I know that this is a few weeks old and I will update  
> and/or raise a bug once I've upgraded, but it would be nice to know  
> if anyone else sees problems with this phone,
>
> regards
> Dave
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 7:50 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF  
> payloadcheck.
>
> It usually means your device can't do rfc2833 correctly and needs to  
> have a bug opened so they can fix it.
>
> /b
>
> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>
>>
>> It appears in response to keypad events during voicemail retrieval,  
>> the prompts were actioned correctly, but I was wondering what the  
>> message actually means ?
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From stevendt at primrosebank.net  Wed Mar  3 12:23:02 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:23:02 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed
	DTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
	<3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
Message-ID: <36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>

Hi Mathieu,

if you tell me how to generate one, I'll do it now ?

regards
Dave
  ----- Original Message ----- 
  From: Mathieu Rene 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:09 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


  I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








  On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


    Hi,

    thanks Brian.

    The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

    regards
    Dave
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 7:50 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


      It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


      /b


      On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



        It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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



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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From stevendt at primrosebank.net  Wed Mar  3 12:39:13 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:39:13 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
Message-ID: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>

Some more info . . . 

The error only seems to occur when a "0" is entered.

Looking at the Phone config, the RFC2833 options are :-

In Band
Out of Band (RFC2833) - currently selected
SIP Info

RTP Payload Type (96-127) - 96 currently selected
RTP DTMP Level (0-63) - 0 Currently selected

Dave
  ----- Original Message ----- 
  From: Dave Stevenson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:23 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


  Hi Mathieu,

  if you tell me how to generate one, I'll do it now ?

  regards
  Dave
    ----- Original Message ----- 
    From: Mathieu Rene 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 03, 2010 8:09 PM
    Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


    I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though. 


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








    On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


      Hi,

      thanks Brian.

      The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

      regards
      Dave
        ----- Original Message -----
        From: Brian West
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 7:50 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


        It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it. 


        /b


        On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



          It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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



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

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


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From chris at fowler.cc  Wed Mar  3 12:48:44 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 3 Mar 2010 15:48:44 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>

Re: Wiki - I'd love to give back to the community;  Does anyone have an objection if I create a "How to use FreeSWITCH on EC2 with RightScale's management dashboard"?

There are some significant differences to the configuration when using RightScale vs. bare bones AWS; and given this is on my employer's time I need to keep it relevant...

The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is a public AMI (you don't need to be a RightScale customer to use it)

Cheers, Chris.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
Sent: Wednesday, March 03, 2010 10:19 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH

Chris Fowler <chris at fowler.cc> said:

> There were no special tricks; you do need to modify/override the
> following with the box's Elastic IP (EIP).
 
This would be helpful to have in the wiki along with any other tips for
your virtual environment. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
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From mrene_lists at avgs.ca  Wed Mar  3 12:51:24 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 15:51:24 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
In-Reply-To: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
	<282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
Message-ID: <D4211946-B75E-4A00-8720-8DAB233E936C@avgs.ca>

You can grab one with wireshark.

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




On 3-Mar-10, at 3:39 PM, Dave Stevenson wrote:

> Some more info . . .
>
> The error only seems to occur when a "0" is entered.
>
> Looking at the Phone config, the RFC2833 options are :-
>
> In Band
> Out of Band (RFC2833) - currently selected
> SIP Info
>
> RTP Payload Type (96-127) - 96 currently selected
> RTP DTMP Level (0-63) - 0 Currently selected
>
> Dave
> ----- Original Message -----
> From: Dave Stevenson
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:23 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196  
> FailedDTMFpayloadcheck.
>
> Hi Mathieu,
>
> if you tell me how to generate one, I'll do it now ?
>
> regards
> Dave
> ----- Original Message -----
> From: Mathieu Rene
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:09 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed  
> DTMFpayloadcheck.
>
> I don't recall any rfc2833 changes in the past 2 weeks. I would like  
> to see a packet capture of those bogus rtp packets though.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:
>
>> Hi,
>>
>> thanks Brian.
>>
>> The device is a Thomson ST2030 which has a basic listing on the  
>> Interop Wiki page, but does not specifically mention rfc2833  
>> (although rtp is marked as "not tested"). I have not seen this  
>> message before while running the FS version that I'm currently  
>> using 16543 - yes, I know that this is a few weeks old and I will  
>> update and/or raise a bug once I've upgraded, but it would be nice  
>> to know if anyone else sees problems with this phone,
>>
>> regards
>> Dave
>> ----- Original Message -----
>> From: Brian West
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Wednesday, March 03, 2010 7:50 PM
>> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF  
>> payloadcheck.
>>
>> It usually means your device can't do rfc2833 correctly and needs  
>> to have a bug opened so they can fix it.
>>
>> /b
>>
>> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>>
>>>
>>> It appears in response to keypad events during voicemail  
>>> retrieval, the prompts were actioned correctly, but I was  
>>> wondering what the message actually means ?
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From stevendt at primrosebank.net  Wed Mar  3 12:56:36 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:56:36 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<D4211946-B75E-4A00-8720-8DAB233E936C@avgs.ca>
Message-ID: <885F4FFA1CFC4A328862377FA9253AE1@bp1.ad.bp.com>

Ahh,

OK, then I'll need to go and do some work on it. WireShark is not installed on the machine. I've never used it, although I do have a copy, so I'll look at installing in on the server and go from there,

regards
Dave
  ----- Original Message ----- 
  From: Mathieu Rene 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:51 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  You can grab one with wireshark.


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








  On 3-Mar-10, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From gmaruzz at celliax.org  Wed Mar  3 13:00:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 3 Mar 2010 22:00:00 +0100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
Message-ID: <7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>

On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
> Re: Wiki - I'd love to give back to the community; ?Does anyone have an objection if I create a "How to use FreeSWITCH on EC2 with RightScale's management dashboard"?
>
> There are some significant differences to the configuration when using RightScale vs. bare bones AWS; and given this is on my employer's time I need to keep it relevant...

Chris,
I know can be more work, but the best would be to let know the
differences too, and how it would be using bare bone AWS. That's also
good for your employer, if he decide to switch back from RightScale...
:)

-giovanni


>
> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is a public AMI (you don't need to be a RightScale customer to use it)
>
> Cheers, Chris.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Wednesday, March 03, 2010 10:19 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>
> Chris Fowler <chris at fowler.cc> said:
>
>> There were no special tricks; you do need to modify/override the
>> following with the box's Elastic IP (EIP).
>
> This would be helpful to have in the wiki along with any other tips for
> your virtual environment. :-)
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rob4manhere at gmail.com  Wed Mar  3 13:09:49 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 3 Mar 2010 15:09:49 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
Message-ID: <25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>

 >  Cheers, Chris.
 >  --
 >  RightScale, Inc.

I doubt they'll be switching any time soon.  I think his employer IS  
RightScale :)


On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:

> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>> Re: Wiki - I'd love to give back to the community;  Does anyone  
>> have an objection if I create a "How to use FreeSWITCH on EC2 with  
>> RightScale's management dashboard"?
>>
>> There are some significant differences to the configuration when  
>> using RightScale vs. bare bones AWS; and given this is on my  
>> employer's time I need to keep it relevant...
>
> Chris,
> I know can be more work, but the best would be to let know the
> differences too, and how it would be using bare bone AWS. That's also
> good for your employer, if he decide to switch back from RightScale...
> :)
>
> -giovanni
>
>
>>
>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is  
>> a public AMI (you don't need to be a RightScale customer to use it)
>>
>> Cheers, Chris.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
>> ] On Behalf Of Russell.Mosemann at cune.org
>> Sent: Wednesday, March 03, 2010 10:19 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>
>> Chris Fowler <chris at fowler.cc> said:
>>
>>> There were no special tricks; you do need to modify/override the
>>> following with the box's Elastic IP (EIP).
>>
>> This would be helpful to have in the wiki along with any other tips  
>> for
>> your virtual environment. :-)
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at celliax.org  Wed Mar  3 13:15:38 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 3 Mar 2010 22:15:38 +0100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com> 
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
Message-ID: <7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>

On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> ?> ?Cheers, Chris.
> ?> ?--
> ?> ?RightScale, Inc.
>
> I doubt they'll be switching any time soon. ?I think his employer IS
> RightScale :)

Oooops, I'll learn to pay attention to signatures :)


>
>
> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>
>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>> Re: Wiki - I'd love to give back to the community; ?Does anyone
>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>> RightScale's management dashboard"?
>>>
>>> There are some significant differences to the configuration when
>>> using RightScale vs. bare bones AWS; and given this is on my
>>> employer's time I need to keep it relevant...
>>
>> Chris,
>> I know can be more work, but the best would be to let know the
>> differences too, and how it would be using bare bone AWS. That's also
>> good for your employer, if he decide to switch back from RightScale...
>> :)
>>
>> -giovanni
>>
>>
>>>
>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>> a public AMI (you don't need to be a RightScale customer to use it)
>>>
>>> Cheers, Chris.
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>> ] On Behalf Of Russell.Mosemann at cune.org
>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>
>>> Chris Fowler <chris at fowler.cc> said:
>>>
>>>> There were no special tricks; you do need to modify/override the
>>>> following with the box's Elastic IP (EIP).
>>>
>>> This would be helpful to have in the wiki along with any other tips
>>> for
>>> your virtual environment. :-)
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From helmut.kuper at ewetel.de  Wed Mar  3 16:20:02 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 04 Mar 2010 01:20:02 +0100
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
References: <4B8EA777.9080207@ewetel.de>
	<0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
Message-ID: <4B8EFCB2.9070709@ewetel.de>

Hi Brian,

does this work for serial calls as well ?


Am 03.03.2010 19:21, schrieb Brian West:
> FreeSWITCH already does this for you on cancel when the call is answered elsewhere.
>
> /b
>
> On Mar 3, 2010, at 12:16 PM, Helmut Kuper wrote:
>
>    
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hello,
>>
>> I'm looking for a way to add a reason header field (like 'Reason:
>> SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
>> send by bridge application when a call times out or was denied.
>>
>> I would like to set it on demand in dialplan, e.g. when a queue is
>> calling a queue member and I don't want to have that call in the phone's
>> missed calls list.
>>
>> Is there a way in FS to do this, yet?
>>
>> best regards
>> helmut
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.7 (MingW32)
>>
>> iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
>> hcWP9lUEFxINGpyLXgfDKLU=
>> =8qjy
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>    



From mgg at giagnocavo.net  Wed Mar  3 16:31:54 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 3 Mar 2010 19:31:54 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670353534F47@mse17be1.mse17.exchange.ms>

Without knowing more about your account and provider, perhaps they have had issues with users getting their accounts hacked, so they added on IP authentication in addition to diget? Or maybe they've had issues with IP-auth people still having DoS type attacks by sending fake INVITEs and using up credit a bit at a time, so they added on digest? Or maybe they're paranoid?

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jonas Gauffin
Sent: Wednesday, March 03, 2010 1:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] OT: SIP and IP spoofing

Hello,

My sip gateway provider are using both IP address locking (only my servers IP addresses can use my gateway account) and Digest authentication on every call.
I asked why and they said that the account would be vulnerable to IP spoofing otherwise. Is that possible? I mean, if someone fakes my servers IP address in the packets, shouldn't the responses be sent back to my server and not the one creating the fake packets? Are there any other reasons to use both ip locking and digest authentication?

Regards,
  Jonas
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From brian at freeswitch.org  Wed Mar  3 16:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 18:32:28 -0600
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <4B8EFCB2.9070709@ewetel.de>
References: <4B8EA777.9080207@ewetel.de>
	<0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
	<4B8EFCB2.9070709@ewetel.de>
Message-ID: <1FF8C930-82A8-4A62-9E4C-503F1FA7AD62@freeswitch.org>

Yes if the call was to say 10 phones at once... and the person answers they hangup with LOSE_RACE which results in the reason header doing that.

in mod_sofia

            if (cause > 0 && cause < 128) {                                                                                                                     
                switch_snprintf(reason, sizeof(reason), "Q.850;cause=%d;text=\"%s\"", cause, switch_channel_cause2str(cause));                                  
            } else if (cause == SWITCH_CAUSE_PICKED_OFF || cause == SWITCH_CAUSE_LOSE_RACE) {
                                                                                   switch_snprintf(reason, sizeof(reason), "SIP;cause=200;text=\"Call completed elsewhere\"");                                                     
            } else {                                                                                                                                            
                switch_snprintf(reason, sizeof(reason), "%s;cause=%d;text=\"%s\"", tech_pvt->profile->username, cause, switch_channel_cause2str(cause));        
            }                                                                                                                                         


/b


On Mar 3, 2010, at 6:20 PM, Helmut Kuper wrote:

> Hi Brian,
> 
> does this work for serial calls as well ?
> 
> 
> Am 03.03.2010 19:21, schrieb Brian West:
>> FreeSWITCH already does this for you on cancel when the call is answered elsewhere.
>> 
>> /b



From mrene_lists at avgs.ca  Wed Mar  3 16:34:51 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 19:34:51 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
Message-ID: <692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>

Theorically you can spoof the network's ip address as long as you set  
the contact header to where the reply should be sent.

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




On 3-Mar-10, at 3:07 AM, Jonas Gauffin wrote:

> Hello,
>
> My sip gateway provider are using both IP address locking (only my  
> servers IP addresses can use my gateway account) and Digest  
> authentication on every call.
> I asked why and they said that the account would be vulnerable to IP  
> spoofing otherwise. Is that possible? I mean, if someone fakes my  
> servers IP address in the packets, shouldn't the responses be sent  
> back to my server and not the one creating the fake packets? Are  
> there any other reasons to use both ip locking and digest  
> authentication?
>
> Regards,
>   Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Wed Mar  3 17:04:08 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 4 Mar 2010 09:04:08 +0800
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
Message-ID: <23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>

we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
pretty well. Of course no much load.

2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>> ?> ?Cheers, Chris.
>> ?> ?--
>> ?> ?RightScale, Inc.
>>
>> I doubt they'll be switching any time soon. ?I think his employer IS
>> RightScale :)
>
> Oooops, I'll learn to pay attention to signatures :)
>
>
>>
>>
>> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>
>>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>>> Re: Wiki - I'd love to give back to the community; ?Does anyone
>>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>>> RightScale's management dashboard"?
>>>>
>>>> There are some significant differences to the configuration when
>>>> using RightScale vs. bare bones AWS; and given this is on my
>>>> employer's time I need to keep it relevant...
>>>
>>> Chris,
>>> I know can be more work, but the best would be to let know the
>>> differences too, and how it would be using bare bone AWS. That's also
>>> good for your employer, if he decide to switch back from RightScale...
>>> :)
>>>
>>> -giovanni
>>>
>>>
>>>>
>>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>>> a public AMI (you don't need to be a RightScale customer to use it)
>>>>
>>>> Cheers, Chris.
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>> ] On Behalf Of Russell.Mosemann at cune.org
>>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>>
>>>> Chris Fowler <chris at fowler.cc> said:
>>>>
>>>>> There were no special tricks; you do need to modify/override the
>>>>> following with the box's Elastic IP (EIP).
>>>>
>>>> This would be helpful to have in the wiki along with any other tips
>>>> for
>>>> your virtual environment. :-)
>>>>
>>>> --
>>>> Russell Mosemann
>>>>
>>>>
>>>>
>>>> ________________________________________________________
>>>> Concordia University, Nebraska
>>>> See http://www.cune.edu/ for the latest news and events!
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mgg at giagnocavo.net  Wed Mar  3 17:25:31 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 3 Mar 2010 20:25:31 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
	<692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670353534F74@mse17be1.mse17.exchange.ms>

Correct me if I'm wrong; it's fortunately been a while since I've had to deal with SIP details. This attack would depend on the strength of the To tag, right? Hopefully they would append ;received to Vias, so the responses would go to the spoofed IP and not the attacker's IP in the Via. The attacker would have to be able to calculate the To tag to construct a valid ACK. 

But that does depend on received being added and secure To tags - hopefully a security-conscious provider would check their side for that.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Wednesday, March 03, 2010 5:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] OT: SIP and IP spoofing

Theorically you can spoof the network's ip address as long as you set the contact header to where the reply should be sent.

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




On 3-Mar-10, at 3:07 AM, Jonas Gauffin wrote:

> Hello,
>
> My sip gateway provider are using both IP address locking (only my 
> servers IP addresses can use my gateway account) and Digest 
> authentication on every call.
> I asked why and they said that the account would be vulnerable to IP 
> spoofing otherwise. Is that possible? I mean, if someone fakes my 
> servers IP address in the packets, shouldn't the responses be sent 
> back to my server and not the one creating the fake packets? Are there 
> any other reasons to use both ip locking and digest authentication?
>
> Regards,
>   Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org


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


From delorenzodesign at gmail.com  Wed Mar  3 19:04:19 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 3 Mar 2010 22:04:19 -0500
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback doesn't seem to get 	called
In-Reply-To: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
Message-ID: <5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>

Has anyone experienced issues with vmd or dtmf with Verizon wireless?  I
have another script that doesn't even register digits pressed -- it's like
no keys are pressed at all.

I've checked the FS log and it doesn't seem to log anything with regard to
the mod_vmd other than an indication of MIN_TIME of 8000 (not sure what that
does exactly).

Does anyone have any sample scripts available other than what's on the FS
wiki?

On Mon, Mar 1, 2010 at 10:17 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I've got the following Lua script working in a sense, but the InputCallback
> never seems to get called while the file is being streamed to the call
> recipient.  I've tried moving the "vmd start" command and set input callback
> around a bit, but to no avail.  I'm testing this against a cell phone voice
> mailbox (Verizon).
>
>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>>  number_to_call = argv[1]
>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>
>> voicemail_detected = false;
>>
>> function onInput(s, type, obj)
>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>> *************\n");
>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>> session:getVariable("vmd_detect") .. " *************\n");
>>
>>      if(type == "event" and voicemail_detected == false) then
>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>> MACHINE DETECTED *************\n");
>>         voicemail_detected = true;
>>         return "break";
>>     end
>> end
>>
>> function playbackMessage()
>>     sleep_time = 1000;
>>     if(voicemail_detected) then
>>        sleep_time = 2500;
>>     end
>>     -- sleep a second
>>     session:sleep(sleep_time);
>>     -- play a file
>>     session:streamFile(message_to_play);
>> end
>>
>> function notify()
>>     session =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>> .. number_to_call)
>>
>>
>>
>>     if(session:ready()) then
>>         -- answer the call
>>         session:answer();
>>         session:setInputCallback("onInput", "");
>>         session:execute("vmd","start");
>>
>>         playbackMessage();
>>         if(voicemail_detected) then
>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>             playbackMessage();
>>         end
>>
>>         freeswitch.consoleLog("notice", "********* hanging up session
>> **********\n");
>>         -- hangup
>>         session:hangup();
>>     end
>> end
>>
>> notify();
>>
>>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>
> --
> Michael De Lorenzo
>
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From christian.bourke1 at gmail.com  Wed Mar  3 19:37:08 2010
From: christian.bourke1 at gmail.com (Bilbo)
Date: Wed, 03 Mar 2010 21:37:08 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1s4-0001px-NS@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2896                           	Bilbo ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From dujinfang at gmail.com  Wed Mar  3 19:37:15 2010
From: dujinfang at gmail.com (seven)
Date: Wed, 03 Mar 2010 21:37:15 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895                           	seven ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From msc at freeswitch.org  Wed Mar  3 19:39:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 03 Mar 2010 21:39:43 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1uZ-0002WX-Tu@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2887                           	Michael Collins ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From kristian.kielhofner at gmail.com  Wed Mar  3 19:41:15 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 03 Mar 2010 21:41:15 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2882                           	Kristian Kielhofner ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From bjbrashier at gmail.com  Wed Mar  3 19:42:05 2010
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 03 Mar 2010 21:42:05 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1wr-0003Er-7J@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2871                           	Bradley Brashier ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From brian at freeswitch.org  Wed Mar  3 20:30:16 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 22:30:16 -0600
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Tienes_una_invitaci=F3n__a_Club?=
	=?iso-8859-1?q?WNC?=
In-Reply-To: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>
References: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>
Message-ID: <1CDF95CD-5331-41E8-8B7B-9073AB6D0F7A@freeswitch.org>

WTF is going on here.

Did someone hit google or something?  REALLY?

This clubwnc people need to be beat with a sledge hammer.

/b



From brian at freeswitch.org  Wed Mar  3 20:47:40 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 22:47:40 -0600
Subject: [Freeswitch-users] TEST
Message-ID: <6C78AD58-771B-41FC-8F93-0D2634C44682@freeswitch.org>

TEST

/b



From christian.bourke1 at gmail.com  Wed Mar  3 20:37:44 2010
From: christian.bourke1 at gmail.com (Bilbo)
Date: Wed, 03 Mar 2010 22:37:44 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn2oi-0000vz-8g@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2896                           	Bilbo ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From infos at madovsky.org  Wed Mar  3 22:09:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:09:30 -0500
Subject: [Freeswitch-users] svn 16892 compilation on Fedora10 64bits
Message-ID: <2A95130D21884F8B988302547CA5AB45@MOBILEE1705>

Hello,

just to say that svn version 16892 compiles
well now and CPU load is ok (xeon)
excellent job !

Regards

F
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From infos at madovsky.org  Wed Mar  3 22:31:35 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:31:35 -0500
Subject: [Freeswitch-users] registration status
Message-ID: <A7363AFFB2024D2A92060998A1A91036@MOBILEE1705>

Is the registration status in DB ?

thanks

Franck
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From infos at madovsky.org  Wed Mar  3 22:33:52 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:33:52 -0500
Subject: [Freeswitch-users] registration status
Message-ID: <E9A73C7FD69D41A58DBEEC6E621F8DC9@MOBILEE1705>

Forget my question I found the answer 

http://wiki.freeswitch.org/wiki/ODBC#ODBC_in_the_core

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 1:31 AM
  Subject: registration status


  Is the registration status in DB ?

  thanks

  Franck
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From dujinfang at gmail.com  Wed Mar  3 22:46:23 2010
From: dujinfang at gmail.com (seven)
Date: Thu, 04 Mar 2010 00:46:23 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4pD-0003Bv-CU@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895                           	seven ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From msc at freeswitch.org  Wed Mar  3 22:47:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 04 Mar 2010 00:47:12 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4q0-0003LZ-K4@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2887                           	Michael Collins ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From kristian.kielhofner at gmail.com  Wed Mar  3 22:47:40 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 04 Mar 2010 00:47:40 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4qS-0003Rz-Nb@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2882                           	Kristian Kielhofner ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From jonas.gauffin at gmail.com  Wed Mar  3 23:01:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 4 Mar 2010 08:01:54 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <4B8E6902.90106@fx-services.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<4B8E6902.90106@fx-services.com>
Message-ID: <d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>

You are correct, I'm using DNSSRV.

imho it's a bug when FS is not using the same nonce in the response as it
received in the request. It might be a design flaw only that only appears
when running against multiple servers through DNSSRV, but it's still a bug.

I'm sure it's easy to fix for Anthony with his m4d sk1llz ;)

On Wed, Mar 3, 2010 at 2:49 PM, Robin Vleij <viper at fx-services.com> wrote:

> On 2010-03-02 16:02, Jonas Gauffin wrote:
>
> Hi Jonas,
>
> > I got a problem with rejected calls due to 403 (happens sometimes).
> > My sip provider found the problem, the nonce used by FS is sometimes
> > incorrect (I got the trace from my sip provider).
>
> Your problem is not a bug. The problem is that your SIP provider has two
> proxies (that you probably find via SRV). Instead of using one gateway
> profile that selects either of them, you have to configure both gateways
> as separate elements. That way FS will use the nonce it gets in the 407,
> instead of mixing them up between the two proxies (who don't have state).
>
> You either use one gateway for outbound and the other one as failover in
> the bridge app, or you use mod_distribute to distribute.
>
> /Robin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Thu Mar  4 00:20:10 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 03:20:10 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com><7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
	<8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>
Message-ID: <4DEA455178544A028AFCA7D21644E42E@MOBILEE1705>

sorry make samples
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 12:52 PM
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings


  Thanks for replying.  I tried make install samples but it didn't work.

   

  My original question should have been "How would an individual user delete a custom voicemail greeting and reset back to the default greeting?"

   

  Thanks,

  Robert

   


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

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Tuesday, March 02, 2010 6:47 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

   

  make install samples

    ----- Original Message ----- 

    From: Robert Hadley 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Tuesday, March 02, 2010 4:31 PM

    Subject: [Freeswitch-users] Delete custom voicemail greetings

     

    Hi FS users,

     

    How do I delete custom voicemail greetings and go back to the default greeting? 

     

    Thanks,

    Robert


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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jaybinks at gmail.com  Thu Mar  4 00:38:27 2010
From: jaybinks at gmail.com (jay binks)
Date: Thu, 4 Mar 2010 18:38:27 +1000
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a difference )
Message-ID: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>

has anyone done any work on getting SNMP monitoring of freeswitch ??

from looking at some other emails about this, it seems that there is not a
heap of desire ( from developers point of view )
to have a FS SNMP module.   Ive also noticed most requests for this type of
monitoring are directed to ESL.

so my thought,  has anyone done any work on building a script / module for
NET-SNMP ( http://net-snmp.sourceforge.net/ )
that uses ESL to fetch its data.

so we would run snmpd on our FS boxes, with the FS-ESL module ( or script )
loaded into snmpd.
from there SNMP requests would be handled by snmpd and the data would then
be pulled over ESL directly from freeswitch.

this seems like a decent solution that would not risk stability of FS by
introducing yet another module.

so I guess my question here is, has anyone done something like this ?
what do you all think of this idea ?
is there anyone who would like to contribute to this idea or offer other
suggestions.

oh yea, and why SNMP..  I duno...
SNMP is just so easy to integrate into other monitoring systems.


-- 
Sincerely

Jay
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From testeador01 at gmail.com  Thu Mar  4 05:57:42 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 4 Mar 2010 08:57:42 -0500
Subject: [Freeswitch-users] TEST
Message-ID: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>

you failed!!  O_O
after the messages from clubWNC, I'm not getting any more list messages on
my e-mail inbox, is it normal?
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From brian at freeswitch.org  Thu Mar  4 06:40:15 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 08:40:15 -0600
Subject: [Freeswitch-users] Testing
Message-ID: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>

Testing

/b



From abrownworth at bandwidth.com  Wed Mar  3 13:53:49 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Wed, 3 Mar 2010 16:53:49 -0500
Subject: [Freeswitch-users] Splitting a Call
Message-ID: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>

Hello fellow FreeSWITCH users,

I'm using FreeSWITCH Version 1.0.trunk (16883) and I'm using the event
socket to break an in-progress call into two independently controlled legs.
I want to be able to work with each leg completely independently - hanging
up one without affecting the other. Per the docs:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_transfer I'm using
uuid_transfer with an inline lua script to work with one leg without hanging
up on the other. I'm setting "hangup_after_bridge" to false on both channels
and invoking uuid_transfer with the -bleg option but as soon as I transfer,
FreeSWITCH hangs up on one of the legs.

Here's what I'm sending on the event socket. (everything gets an "OK"
response from FreeSWITCH)

Set hangup_after_bridge:


api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a hangup_after_bridge
false
api uuid_setvar 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 hangup_after_bridge
false

And then do the uuid_transfer:

api uuid_transfer 53d37581-1f90-44bf-860a-addbc8430e3a -bleg
'lua:callee.lua' inline
api uuid_transfer 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 -bleg
'lua:caller.lua' inline

The callee leg (53d37581-1f90-44bf-860a-addbc8430e3a) is immediately hung up
and the caller leg runs fine. Here's the log messages: (6505551212 is the
caller and 5165551212 is the callee)

2010-03-03 21:29:56.086319 [DEBUG] switch_ivr.c:1441 (sofia/external/+
16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.086319 [NOTICE] switch_ivr.c:1447 Transfer
sofia/external/+16505551212 at 192.168.27.72 to inline[lua:callee.lua at public]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/external/+16505551212 at 192.168.27.72]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_rtp.c:1626 Send middle packet for
[1] ts=91040 dur=1760/1760/2440 seq=7493
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/external/+15165551212]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.092461 [NOTICE] switch_ivr_bridge.c:637 Hangup
sofia/external/+15165551212 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2010-03-03 21:29:56.092461 [DEBUG] switch_channel.c:2071 Send signal
sofia/external/+15165551212 [KILL]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:351
(sofia/external/+15165551212) State EXCHANGE_MEDIA going to sleep
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+15165551212) Running State Change CS_HANGUP
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
(sofia/external/+15165551212) State HANGUP
2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:411 Channel
sofia/external/+15165551212 hanging up, cause: NORMAL_CLEARING
2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/external/+15165551212
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:46
sofia/external/+15165551212 Standard HANGUP, cause: NORMAL_CLEARING
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
(sofia/external/+15165551212) State HANGUP going to sleep
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+15165551212) State Change CS_HANGUP -> CS_REPORTING
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+15165551212) Running State Change CS_REPORTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:590
(sofia/external/+15165551212) State REPORTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING
2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:140 sofia/external/+
16505551212 at 192.168.27.72 SOFIA ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:77
sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:119
(sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
CS_EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:226 sofia/external/+
16505551212 at 192.168.27.72 SOFIA EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:157
sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(callee.lua)
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:53
sofia/external/+15165551212 Standard REPORTING, cause: NORMAL_CLEARING
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:590
(sofia/external/+15165551212) State REPORTING going to sleep
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:327
(sofia/external/+15165551212) State Change CS_REPORTING -> CS_DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1161 Session 24
(sofia/external/+15165551212) Locked, Waiting on external entities
2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1179 Session 24
(sofia/external/+15165551212) Ended
2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/external/+15165551212 [CS_DESTROY]
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:428
(sofia/external/+15165551212) Running State Change CS_DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
(sofia/external/+15165551212) State DESTROY
2010-03-03 21:29:56.095544 [DEBUG] mod_sofia.c:338
sofia/external/+15165551212 SOFIA DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:60
sofia/external/+15165551212 Standard DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
(sofia/external/+15165551212) State DESTROY going to sleep
2010-03-03 21:29:56.127189 [DEBUG] switch_ivr.c:1441 (sofia/external/+
16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.127189 [NOTICE] switch_ivr.c:1447 Transfer
sofia/external/+16505551212 at 192.168.27.72 to inline[lua:caller.lua]
2010-03-03 21:29:56.875077 [DEBUG] switch_core_media_bug.c:360 Attaching BUG
to sofia/external/+16505551212 at 192.168.27.72
2010-03-03 21:29:56.909780 [DEBUG] switch_core_media_bug.c:442 Removing BUG
from sofia/external/+16505551212 at 192.168.27.72
2010-03-03 21:29:56.910802 [ERR] inline:1  Session is not active!
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING
2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:140 sofia/external/+
16505551212 at 192.168.27.72 SOFIA ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:77
sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:119
(sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
CS_EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:226 sofia/external/+
16505551212 at 192.168.27.72 SOFIA EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:157
sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(caller.lua)
2010-03-03 21:29:56.911830 [NOTICE] switch_cpp.cpp:1142 ***********
Caller.lua - Success: 53d37581-1f90-44bf-860a-addbc8430e3a ***********
2010-03-03 21:29:56.911830 [ERR] switch_core_session.c:357 Could not locate
channel type 53d37581-1f90-44bf-860a-addbc8430e3a
2010-03-03 21:29:56.911830 [ERR] switch_ivr_originate.c:2389 Cannot create
outgoing channel of type [53d37581-1f90-44bf-860a-addbc8430e3a] cause:
[CHAN_NOT_IMPLEMENTED]
2010-03-03 21:29:56.911830 [DEBUG] switch_ivr_originate.c:3187 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2010-03-03 21:29:56.911830 [ERR] switch_cpp.cpp:618 session is not
initalized
2010-03-03 21:29:57.032095 [DEBUG] switch_ivr_play_say.c:1156 Codec
Activated L16 at 8000hz 1 channels 20ms

How can I get the callee leg to stay up? Is this not what -bleg is supposed
to do? Am I doing something wrong here? I've tried parking both legs and
invoking the lua on the channels but one of the channels still drops and the
uuid is no-longer available in the event socket.

Is there some other preferred way to split a call apart?

Thanks.

-Anders
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From abrownworth at bandwidth.com  Wed Mar  3 14:56:37 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Wed, 3 Mar 2010 17:56:37 -0500
Subject: [Freeswitch-users] Splitting a Call
In-Reply-To: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>
References: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>
Message-ID: <dbad22fd1003031456k631a4237tea2663cc03ceb776@mail.gmail.com>

Answering my own question, parking the call with "uuid_transfer <uuid> -both
'park' inline" first keeps the other leg busy and things won't hang up.

-Anders

On Wed, Mar 3, 2010 at 4:53 PM, Anders Brownworth <abrownworth at bandwidth.com
> wrote:

> Hello fellow FreeSWITCH users,
>
> I'm using FreeSWITCH Version 1.0.trunk (16883) and I'm using the event
> socket to break an in-progress call into two independently controlled legs.
> I want to be able to work with each leg completely independently - hanging
> up one without affecting the other. Per the docs:
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_transfer I'm using
> uuid_transfer with an inline lua script to work with one leg without hanging
> up on the other. I'm setting "hangup_after_bridge" to false on both channels
> and invoking uuid_transfer with the -bleg option but as soon as I transfer,
> FreeSWITCH hangs up on one of the legs.
>
> Here's what I'm sending on the event socket. (everything gets an "OK"
> response from FreeSWITCH)
>
> Set hangup_after_bridge:
>
>
> api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a hangup_after_bridge
> false
> api uuid_setvar 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 hangup_after_bridge
> false
>
> And then do the uuid_transfer:
>
> api uuid_transfer 53d37581-1f90-44bf-860a-addbc8430e3a -bleg
> 'lua:callee.lua' inline
> api uuid_transfer 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 -bleg
> 'lua:caller.lua' inline
>
> The callee leg (53d37581-1f90-44bf-860a-addbc8430e3a) is immediately hung
> up and the caller leg runs fine. Here's the log messages: (6505551212 is the
> caller and 5165551212 is the callee)
>
> 2010-03-03 21:29:56.086319 [DEBUG] switch_ivr.c:1441 (sofia/external/+
> 16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
> 2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.086319 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/external/+16505551212 at 192.168.27.72 to inline[lua:callee.lua at public]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/external/+16505551212 at 192.168.27.72]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_rtp.c:1626 Send middle packet for
> [1] ts=91040 dur=1760/1760/2440 seq=7493
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/external/+15165551212]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.092461 [NOTICE] switch_ivr_bridge.c:637 Hangup
> sofia/external/+15165551212 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_channel.c:2071 Send signal
> sofia/external/+15165551212 [KILL]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/+15165551212) State EXCHANGE_MEDIA going to sleep
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+15165551212) Running State Change CS_HANGUP
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/+15165551212) State HANGUP
> 2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:411 Channel
> sofia/external/+15165551212 hanging up, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:454 Sending BYE to
> sofia/external/+15165551212
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:46
> sofia/external/+15165551212 Standard HANGUP, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/+15165551212) State HANGUP going to sleep
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+15165551212) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+15165551212) Running State Change CS_REPORTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/+15165551212) State REPORTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:140 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:77
> sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:119
> (sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:226 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:157
> sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
> EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(callee.lua)
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:53
> sofia/external/+15165551212 Standard REPORTING, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/+15165551212) State REPORTING going to sleep
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/+15165551212) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1161 Session 24
> (sofia/external/+15165551212) Locked, Waiting on external entities
> 2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1179 Session 24
> (sofia/external/+15165551212) Ended
> 2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/+15165551212 [CS_DESTROY]
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/+15165551212) Running State Change CS_DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/+15165551212) State DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] mod_sofia.c:338
> sofia/external/+15165551212 SOFIA DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:60
> sofia/external/+15165551212 Standard DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/+15165551212) State DESTROY going to sleep
> 2010-03-03 21:29:56.127189 [DEBUG] switch_ivr.c:1441 (sofia/external/+
> 16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
> 2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.127189 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/external/+16505551212 at 192.168.27.72 to inline[lua:caller.lua]
> 2010-03-03 21:29:56.875077 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/external/+16505551212 at 192.168.27.72
> 2010-03-03 21:29:56.909780 [DEBUG] switch_core_media_bug.c:442 Removing BUG
> from sofia/external/+16505551212 at 192.168.27.72
> 2010-03-03 21:29:56.910802 [ERR] inline:1  Session is not active!
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:140 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:77
> sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:119
> (sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:226 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:157
> sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
> EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(caller.lua)
> 2010-03-03 21:29:56.911830 [NOTICE] switch_cpp.cpp:1142 ***********
> Caller.lua - Success: 53d37581-1f90-44bf-860a-addbc8430e3a ***********
> 2010-03-03 21:29:56.911830 [ERR] switch_core_session.c:357 Could not locate
> channel type 53d37581-1f90-44bf-860a-addbc8430e3a
> 2010-03-03 21:29:56.911830 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [53d37581-1f90-44bf-860a-addbc8430e3a] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2010-03-03 21:29:56.911830 [DEBUG] switch_ivr_originate.c:3187 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2010-03-03 21:29:56.911830 [ERR] switch_cpp.cpp:618 session is not
> initalized
> 2010-03-03 21:29:57.032095 [DEBUG] switch_ivr_play_say.c:1156 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> How can I get the callee leg to stay up? Is this not what -bleg is supposed
> to do? Am I doing something wrong here? I've tried parking both legs and
> invoking the lua on the channels but one of the channels still drops and the
> uuid is no-longer available in the event socket.
>
> Is there some other preferred way to split a call apart?
>
> Thanks.
>
> -Anders
>
>
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From brian at freeswitch.org  Thu Mar  4 06:52:01 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 08:52:01 -0600
Subject: [Freeswitch-users] SUCCESS!!!
Message-ID: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>

I have to say PYTHON SUCKS... ie Mailman is evil.

/b



From testa at voicetechnology.com.br  Thu Mar  4 06:55:24 2010
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Thu, 4 Mar 2010 11:55:24 -0300
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
Message-ID: <9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>

I used to do tests on conferencing using linode.com, which is Xen VMs
without issues, although there were not much load on it.

On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:

> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
> pretty well. Of course no much load.
>
> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
> wrote:
> >>  >  Cheers, Chris.
> >>  >  --
> >>  >  RightScale, Inc.
> >>
> >> I doubt they'll be switching any time soon.  I think his employer IS
> >> RightScale :)
> >
> > Oooops, I'll learn to pay attention to signatures :)
> >
> >
> >>
> >>
> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
> >>
> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
> >>>> RightScale's management dashboard"?
> >>>>
> >>>> There are some significant differences to the configuration when
> >>>> using RightScale vs. bare bones AWS; and given this is on my
> >>>> employer's time I need to keep it relevant...
> >>>
> >>> Chris,
> >>> I know can be more work, but the best would be to let know the
> >>> differences too, and how it would be using bare bone AWS. That's also
> >>> good for your employer, if he decide to switch back from RightScale...
> >>> :)
> >>>
> >>> -giovanni
> >>>
> >>>
> >>>>
> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
> >>>> a public AMI (you don't need to be a RightScale customer to use it)
> >>>>
> >>>> Cheers, Chris.
> >>>>
> >>>> -----Original Message-----
> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> >>>> ] On Behalf Of Russell.Mosemann at cune.org
> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
> >>>> To: freeswitch-users at lists.freeswitch.org
> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
> >>>>
> >>>> Chris Fowler <chris at fowler.cc> said:
> >>>>
> >>>>> There were no special tricks; you do need to modify/override the
> >>>>> following with the box's Elastic IP (EIP).
> >>>>
> >>>> This would be helpful to have in the wiki along with any other tips
> >>>> for
> >>>> your virtual environment. :-)
> >>>>
> >>>> --
> >>>> Russell Mosemann
> >>>>
> >>>>
> >>>>
> >>>> ________________________________________________________
> >>>> Concordia University, Nebraska
> >>>> See http://www.cune.edu/ for the latest news and events!
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>> Sincerely,
> >>>
> >>> Giovanni Maruzzelli
> >>> Cell : +39-347-2665618
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mustafa.pk at gmail.com  Thu Mar  4 07:08:34 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 4 Mar 2010 20:08:34 +0500
Subject: [Freeswitch-users] Testing
In-Reply-To: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>
References: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>
Message-ID: <8213d6071003040708i790de267g8863980c2d7e0f5f@mail.gmail.com>

echo 123

On Thu, Mar 4, 2010 at 7:40 PM, Brian West <brian at freeswitch.org> wrote:
> Testing
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com


From paul at apcl.us  Thu Mar  4 07:11:33 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 04 Mar 2010 10:11:33 -0500
Subject: [Freeswitch-users] how to see registration info for a particular
	user?
Message-ID: <4B8FCDA5.6000700@apcl.us>

I am familiar with the console command:
     sofia status profile internal reg
to see the registration info for all currently registered users.

Is there a modification to this command that will limit the result to 
just one specific user registered user?

   Thanks,
   Paul



From gmaruzz at celliax.org  Thu Mar  4 07:22:01 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 4 Mar 2010 16:22:01 +0100
Subject: [Freeswitch-users] Your Future Order with 75% off retail
Message-ID: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>

Testing

:P

-giovanni
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From brian at freeswitch.org  Thu Mar  4 07:27:56 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 09:27:56 -0600
Subject: [Freeswitch-users] Your Future Order with 75% off retail
In-Reply-To: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>
References: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>
Message-ID: <EB62BDFE-D71B-4A3A-BF09-CB285EEE9BB7@freeswitch.org>

You crazy!

/b
PS: mirror image of me.

On Mar 4, 2010, at 9:22 AM, Giovanni Maruzzelli wrote:

> Testing
> 
> :P
> 
> -giovanni
> -- 
> Sincerely,



From infos at madovsky.org  Thu Mar  4 08:49:00 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 11:49:00 -0500
Subject: [Freeswitch-users] SUCCESS!!!
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
Message-ID: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>

Mailman, I ll get your scalp 

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 04, 2010 9:52 AM
Subject: [Freeswitch-users] SUCCESS!!!


>I have to say PYTHON SUCKS... ie Mailman is evil.
> 
> /b
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Thu Mar  4 09:13:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 11:13:59 -0600
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/ESL-33
Message-ID: <78042E0E-4E1B-41A9-885F-867DD621FCE2@freeswitch.org>

Any java web heads out there wanna take a smack at this one?

/b



From anthony.minessale at gmail.com  Thu Mar  4 09:17:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 11:17:38 -0600
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
Message-ID: <191c3a031003040917p47f87a73hc559a5d8633b31fc@mail.gmail.com>

Let's upgrade to Angry Dog


On Thu, Mar 4, 2010 at 10:49 AM, Madovsky <infos at madovsky.org> wrote:

> Mailman, I ll get your scalp
>
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, March 04, 2010 9:52 AM
> Subject: [Freeswitch-users] SUCCESS!!!
>
>
> >I have to say PYTHON SUCKS... ie Mailman is evil.
> >
> > /b
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From m.sobkow at marketelsystems.com  Thu Mar  4 09:22:07 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 11:22:07 -0600
Subject: [Freeswitch-users] How to play back a .wav file from Erlang?
Message-ID: <4B8FEC3F.5060102@marketelsystems.com>

I have the UUID of a parked call.  I've waited n seconds for an operator 
to answer that call, but none are available so I want to play a message 
to the customer.  How do I do that from Erlang?  I've tried the 
following, but it doesn't seem to actually play the file, and I'm not 
getting any useful information in fs_cli about the playback attempt that 
would indicate why it's not working.

bgapi( eval, "uuid:1f6d7a61-fd1a-4efd-bf84-31c4eea652c2 playback 
/opt/freeswitch/sounds/en/us/callie/misc/8000/sorry.wav" )

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



From m.sobkow at marketelsystems.com  Thu Mar  4 10:36:23 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 12:36:23 -0600
Subject: [Freeswitch-users] Call keeps hanging up
Message-ID: <4B8FFDA7.50708@marketelsystems.com>

I'm having some difficulty with Freeswitch/Erlang processing.  An 
operator dials in and registers by entering a PIN code, which triggers 
an Erlang event handler callback and puts the call into park.  This works.

A customer calls in, which triggers another Erlang event handler 
callback and puts the call into park.  That works, too.

Another thread of Erlang checks the operator and customer queues, and 
uses uuid_bridge to join the two calls together.  That also works.

What _doesn't_ work is that when the customer hangs up, I want the 
operator A leg to go back into a park state, and it's hanging up 
instead.  One suspicion I have is that because there is no dialplan 
associated with the call (it's controlled by Erlang), there is no 
dialplan to "continue" by setting hangup_after_bridge=false.  Either 
that or maybe I need to set the variables on the new UUID returned by 
the uuid_bridge command (though I thought that UUID was just the UUID 
assigned to the command, not a new call identifier.)

BTW, "pbx" is just an OTP rewrite of pieces of the standard 
"freeswitch.erl" code.


    pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
++ " hangup_after_bridge false" ),
    pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
++ " park_after_bridge true" ),

    BridgeResult = pbx:api( uuid_bridge, 
Operator#pbx_operator_registry.operator_uuid ++ " " ++ 
Customer#pbx_cust_queue.uuid ),

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



From andrew at hijacked.us  Thu Mar  4 10:47:55 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Mar 2010 13:47:55 -0500
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B8FFDA7.50708@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
Message-ID: <20100304184755.GK1751@hijacked.us>

On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
> I'm having some difficulty with Freeswitch/Erlang processing.  An 
> operator dials in and registers by entering a PIN code, which triggers 
> an Erlang event handler callback and puts the call into park.  This works.
> 
> A customer calls in, which triggers another Erlang event handler 
> callback and puts the call into park.  That works, too.
> 
> Another thread of Erlang checks the operator and customer queues, and 
> uses uuid_bridge to join the two calls together.  That also works.
> 
> What _doesn't_ work is that when the customer hangs up, I want the 
> operator A leg to go back into a park state, and it's hanging up 
> instead.  One suspicion I have is that because there is no dialplan 
> associated with the call (it's controlled by Erlang), there is no 
> dialplan to "continue" by setting hangup_after_bridge=false.  Either 
> that or maybe I need to set the variables on the new UUID returned by 
> the uuid_bridge command (though I thought that UUID was just the UUID 
> assigned to the command, not a new call identifier.)
> 
> BTW, "pbx" is just an OTP rewrite of pieces of the standard 
> "freeswitch.erl" code.
> 
> 
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
> ++ " hangup_after_bridge false" ),
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
> ++ " park_after_bridge true" ),
>

Try using park_after_bridge and catch the park event or
transfer_after_bridge and set the dialplan location to transfer the call
to.

Andrew


From anthony.minessale at gmail.com  Thu Mar  4 11:12:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 13:12:50 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>

"Hello this is the United States Calling are we reaching... *click* See!, he
keeps hanging up, is there supposed to be someone there besides your wife
sir?"


On Thu, Mar 4, 2010 at 12:47 PM, Andrew Thompson <andrew at hijacked.us> wrote:

> On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
> > I'm having some difficulty with Freeswitch/Erlang processing.  An
> > operator dials in and registers by entering a PIN code, which triggers
> > an Erlang event handler callback and puts the call into park.  This
> works.
> >
> > A customer calls in, which triggers another Erlang event handler
> > callback and puts the call into park.  That works, too.
> >
> > Another thread of Erlang checks the operator and customer queues, and
> > uses uuid_bridge to join the two calls together.  That also works.
> >
> > What _doesn't_ work is that when the customer hangs up, I want the
> > operator A leg to go back into a park state, and it's hanging up
> > instead.  One suspicion I have is that because there is no dialplan
> > associated with the call (it's controlled by Erlang), there is no
> > dialplan to "continue" by setting hangup_after_bridge=false.  Either
> > that or maybe I need to set the variables on the new UUID returned by
> > the uuid_bridge command (though I thought that UUID was just the UUID
> > assigned to the command, not a new call identifier.)
> >
> > BTW, "pbx" is just an OTP rewrite of pieces of the standard
> > "freeswitch.erl" code.
> >
> >
> >     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> > ++ " hangup_after_bridge false" ),
> >     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> > ++ " park_after_bridge true" ),
> >
>
> Try using park_after_bridge and catch the park event or
> transfer_after_bridge and set the dialplan location to transfer the call
> to.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Thu Mar  4 12:19:40 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 12:19:40 -0800
Subject: [Freeswitch-users] Dialing **<extension>
Message-ID: <613B61547E7C43FFB4978276BDD146D9@greyhawk.tonecommander.com>

I used to be able to dial "**<extension>" to pickup a ringing call (v.
1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
this no longer works the same.

Is this feature still available under some other star-code?

Best Regards,
Jerry



From tculjaga at gmail.com  Thu Mar  4 12:44:23 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 4 Mar 2010 21:44:23 +0100
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
Message-ID: <65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>

On Thu, Mar 4, 2010 at 9:38 AM, jay binks <jaybinks at gmail.com> wrote:

> has anyone done any work on getting SNMP monitoring of freeswitch ??
>
> from looking at some other emails about this, it seems that there is not a
> heap of desire ( from developers point of view )
> to have a FS SNMP module.   Ive also noticed most requests for this type of
> monitoring are directed to ESL.
>

because there is no need :)


>
> so my thought,  has anyone done any work on building a script / module for
> NET-SNMP ( http://net-snmp.sourceforge.net/ )
> that uses ESL to fetch its data.
>
>
what i'm doing is just make a script to call fs_cli -x command

e.g.

tculjaga at nemesis:~$ /usr/local/freeswitch/bin/fs_cli -x "status"
UP 0 years, 0 days, 0 hours, 0 minutes, 8 seconds, 470 milliseconds, 62
microseconds
0 session(s) since startup
0 session(s) 0/30
1000 session(s) max


and bind it to an OID...


I'm even firing SIPP with to check end-to-end FS sanity.



> so we would run snmpd on our FS boxes, with the FS-ESL module ( or script )
> loaded into snmpd.
> from there SNMP requests would be handled by snmpd and the data would then
> be pulled over ESL directly from freeswitch.
>
> this seems like a decent solution that would not risk stability of FS by
> introducing yet another module.
>
> so I guess my question here is, has anyone done something like this ?
>

yes...





> what do you all think of this idea ?
> is there anyone who would like to contribute to this idea or offer other
> suggestions.
>
>
it is just a script ... quite easy to setup...

prepare your script:

[tculjaga@ bin]$ cat activeChannels.sh
#!/bin/sh

CHANNELS=`/usr/local/freeswitch/bin/fs_cli -x "show channels count"
|/bin/grep total | /bin/awk {'print $1'}`
/bin/echo $CHANNELS
exit $CHANNELS
[tculjaga at l01sipindir2 bin]$
[tculjaga at l01sipindir2 bin]$
[tculjaga at l01sipindir2 bin]$ cat currentCallsPerSec.sh
#!/bin/sh



in snmpd.conf add this:

exec activeChannels /usr/local/bin/activeChannels.sh
exec checkCallFlowCPS /usr/local/bin/sipgen/sipgen.sh 1234 1 10




> oh yea, and why SNMP..  I duno...
> SNMP is just so easy to integrate into other monitoring systems.
>
>
because every NMS uses it .. maybe thats why :)


>
>

> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Thu Mar  4 12:48:06 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 14:48:06 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <4B901C86.7040806@marketelsystems.com>

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From jimthomasembedded at yahoo.com  Thu Mar  4 13:10:55 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Thu, 4 Mar 2010 13:10:55 -0800 (PST)
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
Message-ID: <43146.92621.qm@web44811.mail.sp1.yahoo.com>

> PYTHON SUCKS

That's like saying:

? Asterisk sucks.
? Asterisk is written in C language.
? Therefore, C language sucks.

:-)

?
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 04, 2010 9:52 AM
Subject: [Freeswitch-users] SUCCESS!!!


>I have to say PYTHON SUCKS... ie Mailman is evil.
> 
> /b



      
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From brian at freeswitch.org  Thu Mar  4 13:22:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 15:22:17 -0600
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <43146.92621.qm@web44811.mail.sp1.yahoo.com>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
	<43146.92621.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <B9F8B4A9-368B-42D5-B30C-E49756710264@freeswitch.org>

You can create shitty art with paint... doesn't mean the paint sucks.

/b

On Mar 4, 2010, at 3:10 PM, Jim Thomas wrote:

> > PYTHON SUCKS
>  
> That's like saying:
>  
>   Asterisk sucks.
>   Asterisk is written in C language.
>   Therefore, C language sucks.
>  
> :-)
> 

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From christian.loeschenkohl at xpirio.com  Thu Mar  4 13:43:31 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Mar 2010 22:43:31 +0100
Subject: [Freeswitch-users] how to see registration info for a
 particular user?
In-Reply-To: <4B8FCDA5.6000700@apcl.us>
References: <4B8FCDA5.6000700@apcl.us>
Message-ID: <4B902983.9080207@xpirio.com>

hi

simply add the user id to the end of the command
like "sofia status profile internal reg 123456"
where 123456 is the id of the user you want to see

br

Paul Levin wrote:

> I am familiar with the console command:
>      sofia status profile internal reg
> to see the registration info for all currently registered users.
> 
> Is there a modification to this command that will limit the result to 
> just one specific user registered user?
> 
>    Thanks,
>    Paul
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From jerry.richards at teotech.com  Thu Mar  4 14:14:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 14:14:30 -0800
Subject: [Freeswitch-users] Dialing **<extension>
Message-ID: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>


By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry




-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Thursday, March 04, 2010 12:20 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Dialing **<extension>

I used to be able to dial "**<extension>" to pickup a ringing call (v.
1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
this no longer works the same.

Is this feature still available under some other star-code?

Best Regards,
Jerry



From jaybinks at gmail.com  Thu Mar  4 14:25:57 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 08:25:57 +1000
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com> 
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
Message-ID: <ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>

HUH, well there you go..   exactly what I was after...
someone has done exactly what I was thinking.

I know you've sent it to me by email, but lets get this in the wiki.
Ill put this up from your email, but if you have more id encourage you to
share what youve done.

Jay


On Fri, Mar 5, 2010 at 6:44 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Thu, Mar 4, 2010 at 9:38 AM, jay binks <jaybinks at gmail.com> wrote:
>
>> has anyone done any work on getting SNMP monitoring of freeswitch ??
>>
>> from looking at some other emails about this, it seems that there is not a
>> heap of desire ( from developers point of view )
>> to have a FS SNMP module.   Ive also noticed most requests for this type
>> of monitoring are directed to ESL.
>>
>
> because there is no need :)
>
>
>>
>> so my thought,  has anyone done any work on building a script / module for
>> NET-SNMP ( http://net-snmp.sourceforge.net/ )
>> that uses ESL to fetch its data.
>>
>>
> what i'm doing is just make a script to call fs_cli -x command
>
> e.g.
>
> tculjaga at nemesis:~$ /usr/local/freeswitch/bin/fs_cli -x "status"
> UP 0 years, 0 days, 0 hours, 0 minutes, 8 seconds, 470 milliseconds, 62
> microseconds
> 0 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
>
>
> and bind it to an OID...
>
>
> I'm even firing SIPP with to check end-to-end FS sanity.
>
>
>
>> so we would run snmpd on our FS boxes, with the FS-ESL module ( or script
>> ) loaded into snmpd.
>> from there SNMP requests would be handled by snmpd and the data would then
>> be pulled over ESL directly from freeswitch.
>>
>> this seems like a decent solution that would not risk stability of FS by
>> introducing yet another module.
>>
>> so I guess my question here is, has anyone done something like this ?
>>
>
> yes...
>
>
>
>
>
>> what do you all think of this idea ?
>> is there anyone who would like to contribute to this idea or offer other
>> suggestions.
>>
>>
> it is just a script ... quite easy to setup...
>
> prepare your script:
>
> [tculjaga@ bin]$ cat activeChannels.sh
> #!/bin/sh
>
> CHANNELS=`/usr/local/freeswitch/bin/fs_cli -x "show channels count"
> |/bin/grep total | /bin/awk {'print $1'}`
> /bin/echo $CHANNELS
> exit $CHANNELS
> [tculjaga at l01sipindir2 bin]$
> [tculjaga at l01sipindir2 bin]$
> [tculjaga at l01sipindir2 bin]$ cat currentCallsPerSec.sh
> #!/bin/sh
>
>
>
> in snmpd.conf add this:
>
> exec activeChannels /usr/local/bin/activeChannels.sh
> exec checkCallFlowCPS /usr/local/bin/sipgen/sipgen.sh 1234 1 10
>
>
>
>
>> oh yea, and why SNMP..  I duno...
>> SNMP is just so easy to integrate into other monitoring systems.
>>
>>
> because every NMS uses it .. maybe thats why :)
>
>
>>
>>
>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From m.sobkow at marketelsystems.com  Thu Mar  4 14:35:32 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 16:35:32 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <4B9035B4.6030901@marketelsystems.com>

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From msc at freeswitch.org  Thu Mar  4 14:58:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 14:58:36 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>

On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, it appears that the "**<extension>" ringing call pickup feature
> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
> of
> the following scenario:
>
> 1) 1059 calls 5381
> 2) 5381 rings
> 3) 5402 dials "**5381"
>
> In the bad case (http://pastebin.freeswitch.org/12325), all calls
> disconnect
> following above scenario.  In the good case
> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>
> Best Regards,
> Jerry
>

I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC
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From gorand at donevtechconsulting.com  Thu Mar  4 14:59:37 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 16:59:37 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0ca201cabbee$5ee077a0$1ca166e0$@com>

Howdy Freeswitch community; 

I was wondering if the developers are any closer to the release of 1.05
version. Please don't point me to an e-mail link :) Just inquiring as to
whether there is a firm close date of 1.05. 

Thanks. 
Goran



From msc at freeswitch.org  Thu Mar  4 15:00:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:00:12 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B9035B4.6030901@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
Message-ID: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>

On Thu, Mar 4, 2010 at 2:35 PM, Mark Sobkow <m.sobkow at marketelsystems.com>wrote:

>  Andrew Thompson wrote:
>
> On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
>
>
>  I'm having some difficulty with Freeswitch/Erlang processing.  An
> operator dials in and registers by entering a PIN code, which triggers
> an Erlang event handler callback and puts the call into park.  This works.
>
> A customer calls in, which triggers another Erlang event handler
> callback and puts the call into park.  That works, too.
>
> Another thread of Erlang checks the operator and customer queues, and
> uses uuid_bridge to join the two calls together.  That also works.
>
> What _doesn't_ work is that when the customer hangs up, I want the
> operator A leg to go back into a park state, and it's hanging up
> instead.  One suspicion I have is that because there is no dialplan
> associated with the call (it's controlled by Erlang), there is no
> dialplan to "continue" by setting hangup_after_bridge=false.  Either
> that or maybe I need to set the variables on the new UUID returned by
> the uuid_bridge command (though I thought that UUID was just the UUID
> assigned to the command, not a new call identifier.)
>
> BTW, "pbx" is just an OTP rewrite of pieces of the standard
> "freeswitch.erl" code.
>
>
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> ++ " hangup_after_bridge false" ),
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> ++ " park_after_bridge true" ),
>
>
>
>  Try using park_after_bridge and catch the park event or
> transfer_after_bridge and set the dialplan location to transfer the call
> to.
>
> Andrew
>
>
>
>  What a relief.  "transfer_after_bridge" worked!  Plus it provided an
> obvious way for me to collect the call result code, fire it back to Erlang,
> and leave the operator waiting for the next incoming call. :)
>
> "Another satisfied customer!" :D Now it's time to throw some of that
Canadian colored money at Tony's paypal account. ;)
BTW, what kind of solution are you developing?
-MC
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From freeswitch at cartissolutions.com  Thu Mar  4 15:04:26 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:04:26 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>	<20100304184755.GK1751@hijacked.us>
	<191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>
Message-ID: <4B903C7A.7030902@cartissolutions.com>


On 03/04/2010 01:12 PM, Anthony Minessale wrote:
> "Hello this is the United States Calling are we reaching... *click* 
> See!, he keeps hanging up, is there supposed to be someone there 
> besides your wife sir?"
>
>

"And it's a man answering..."   *ring ring...    ring ring...*


From anthony.minessale at gmail.com  Thu Mar  4 15:06:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:06:19 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
Message-ID: <191c3a031003041506r4bad1d75pd28839b3c334dea1@mail.gmail.com>

1) You are supposed to report bugs to jira http://jira.freeswitch.org
2) You are not using svn trunk so the line numbers don't match for
debugging.
3) #2 is a prerequisite for #1


On Thu, Mar 4, 2010 at 4:14 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, it appears that the "**<extension>" ringing call pickup feature
> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
> of
> the following scenario:
>
> 1) 1059 calls 5381
> 2) 5381 rings
> 3) 5402 dials "**5381"
>
> In the bad case (http://pastebin.freeswitch.org/12325), all calls
> disconnect
> following above scenario.  In the good case
> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>
> Best Regards,
> Jerry
>
>
>
>
> -----Original Message-----
> From: Jerry Richards [mailto:jerry.richards at teotech.com]
> Sent: Thursday, March 04, 2010 12:20 PM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: Dialing **<extension>
>
> I used to be able to dial "**<extension>" to pickup a ringing call (v.
> 1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
> this no longer works the same.
>
> Is this feature still available under some other star-code?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar  4 15:07:34 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:07:34 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <4B903D36.2040407@cartissolutions.com>


On 03/04/2010 04:59 PM, Goran Donev wrote:
> Howdy Freeswitch community;
>
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05.
>
> Thanks.
> Goran
>    

"We are delighted that your curiosity of our planet continues unabated.  
And we're like to assure you that the two nuclear-armed missiles are 
merely a courtesy detail..."

Standard response:  It's in the oven baking right now.  The muffin man 
will take it out as soon as it's ready...


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com




From brian at freeswitch.org  Thu Mar  4 15:07:58 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:07:58 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <C278735C-3EA9-46B0-835F-1AF0E241DC34@freeswitch.org>

Its getting there.  Please see jira... and keep an eye out the best thing anyone can do at this point is TEST TEST TEST TEST.  ;)

/b

On Mar 4, 2010, at 4:59 PM, Goran Donev wrote:

> Howdy Freeswitch community; 
> 
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05. 
> 
> Thanks. 
> Goran
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Mar  4 15:09:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:09:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>

No we have no deadline, we have spent countless hours preparing and we still
have a small list of packaging concerns.  Feel free to download the code
from http://latest.freeswitch.org and rest assured it *is* 1.0.5

On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> Howdy Freeswitch community;
>
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05.
>
> Thanks.
> Goran
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar  4 15:15:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:15:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
Message-ID: <191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>

You know, that's the only question you have asked on this list.

I went back and searched for your name.

And you have asked it like 12 times and have been answered the same each
time.

What is the obsession with an arbitrary tag?


On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> No we have no deadline, we have spent countless hours preparing and we
> still have a small list of packaging concerns.  Feel free to download the
> code from http://latest.freeswitch.org and rest assured it *is* 1.0.5
>
> On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <
> gorand at donevtechconsulting.com> wrote:
>
>> Howdy Freeswitch community;
>>
>> I was wondering if the developers are any closer to the release of 1.05
>> version. Please don't point me to an e-mail link :) Just inquiring as to
>> whether there is a firm close date of 1.05.
>>
>> Thanks.
>> Goran
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar  4 15:30:23 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:30:23 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>	<0ca201cabbee$5ee077a0$1ca166e0$@com>	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
Message-ID: <4B90428F.1000408@cartissolutions.com>


On 03/04/2010 05:15 PM, Anthony Minessale wrote:
> You know, that's the only question you have asked on this list.
>
> I went back and searched for your name.
>
> And you have asked it like 12 times and have been answered the same 
> each time.
>
> What is the obsession with an arbitrary tag?

Maybe we should name the 1.0.5 official release as the "Donev Release"...


>
> On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     No we have no deadline, we have spent countless hours preparing
>     and we still have a small list of packaging concerns.  Feel free
>     to download the code from http://latest.freeswitch.org and rest
>     assured it *is* 1.0.5
>
>     On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev
>     <gorand at donevtechconsulting.com
>     <mailto:gorand at donevtechconsulting.com>> wrote:
>
>         Howdy Freeswitch community;
>
>         I was wondering if the developers are any closer to the
>         release of 1.05
>         version. Please don't point me to an e-mail link :) Just
>         inquiring as to
>         whether there is a firm close date of 1.05.
>
>         Thanks.
>         Goran
>
>


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com


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From andrew at hijacked.us  Thu Mar  4 15:40:50 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Mar 2010 18:40:50 -0500
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
Message-ID: <20100304234050.GL1751@hijacked.us>

On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:

> > "Another satisfied customer!" :D Now it's time to throw some of that
> Canadian colored money at Tony's paypal account. ;)

Hey! I wrote the erlang module and provided the solution to the problem
;)

Andrew


From gorand at donevtechconsulting.com  Thu Mar  4 15:41:17 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 17:41:17 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 47
In-Reply-To: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0cd601cabbf4$30f000d0$92d00270$@com>

It has been baking for a while. What's the timer on the oven say, how much
more till it's done.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
freeswitch-users-request at lists.freeswitch.org
Sent: Thursday, March 04, 2010 5:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 45, Issue 47

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of FreeSWITCH-users digest..."



From brian at freeswitch.org  Thu Mar  4 15:41:21 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:41:21 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B90428F.1000408@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>	<0ca201cabbee$5ee077a0$1ca166e0$@com>	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
Message-ID: <12464591-4662-494A-AC23-ED322F8F6B94@freeswitch.org>

HAHAHA NO! :P

/b

On Mar 4, 2010, at 5:30 PM, Yossi Neiman wrote:

> Maybe we should name the 1.0.5 official release as the "Donev Release"...



From jaybinks at gmail.com  Thu Mar  4 15:42:25 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 09:42:25 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B903D36.2040407@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
Message-ID: <ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>

who is the muffin man !??

yuuummm muffins... I Want some...
fresh yummy muffins...

is BKW the muffin man ?

J


> Standard response:  It's in the oven baking right now.  The muffin man
> will take it out as soon as it's ready...
>
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From msc at freeswitch.org  Thu Mar  4 15:53:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:53:45 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 47
In-Reply-To: <0cd601cabbf4$30f000d0$92d00270$@com>
References: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
	<0cd601cabbf4$30f000d0$92d00270$@com>
Message-ID: <87f2f3b91003041553o1309f27dtf81573e6613775f2@mail.gmail.com>

On Thu, Mar 4, 2010 at 3:41 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> It has been baking for a while. What's the timer on the oven say, how much
> more till it's done.
>
> The answer is.... "It depends." For many of us, current SVN is already
baked and ready for action. I've been using SVN on several machines for the
past month and all has been well. For me, SVN has been "1.0.5" for at least
four weeks. The stuff that they're working on is really oddball bugs from
goofy use cases and dealing with bugs from phone manufacturers...

-MC
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From msc at freeswitch.org  Thu Mar  4 15:54:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:54:23 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304234050.GL1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<20100304234050.GL1751@hijacked.us>
Message-ID: <87f2f3b91003041554y1c7fbe51h956b525ab6dcee67@mail.gmail.com>

On Thu, Mar 4, 2010 at 3:40 PM, Andrew Thompson <andrew at hijacked.us> wrote:

> On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:
>
> > > "Another satisfied customer!" :D Now it's time to throw some of that
> > Canadian colored money at Tony's paypal account. ;)
>
> Hey! I wrote the erlang module and provided the solution to the problem
> ;)
>

ok ok. what's your paypal account? We'll give you some love too. :)
-MC
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From brian at freeswitch.org  Thu Mar  4 15:54:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:54:37 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
	<ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>
Message-ID: <E8536232-8A23-4A6D-A3C6-D4F68A28100E@freeswitch.org>

Yes, I know the muffin man. W-who lives on Drewery Lane? 

/b

On Mar 4, 2010, at 5:42 PM, jay binks wrote:

> who is the muffin man !??
> 
> yuuummm muffins... I Want some...
> fresh yummy muffins...
> 
> is BKW the muffin man ?
> 
> J
>  

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From jaybinks at gmail.com  Thu Mar  4 15:54:29 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 09:54:29 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B90428F.1000408@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com> 
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com> 
	<4B90428F.1000408@cartissolutions.com>
Message-ID: <ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>

check this out ...
http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2

<http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2>I
think Goran Donev is Mr "Are we there yet" :)

J

On Fri, Mar 5, 2010 at 9:30 AM, Yossi Neiman <freeswitch at cartissolutions.com
> wrote:

>
> On 03/04/2010 05:15 PM, Anthony Minessale wrote:
>
> You know, that's the only question you have asked on this list.
>
> I went back and searched for your name.
>
> And you have asked it like 12 times and have been answered the same each
> time.
>
> What is the obsession with an arbitrary tag?
>
>
> Maybe we should name the 1.0.5 official release as the "Donev Release"...
>
>
>
>
> On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> No we have no deadline, we have spent countless hours preparing and we
>> still have a small list of packaging concerns.  Feel free to download the
>> code from http://latest.freeswitch.org and rest assured it *is* 1.0.5
>>
>> On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <
>> gorand at donevtechconsulting.com> wrote:
>>
>>> Howdy Freeswitch community;
>>>
>>> I was wondering if the developers are any closer to the release of 1.05
>>> version. Please don't point me to an e-mail link :) Just inquiring as to
>>> whether there is a firm close date of 1.05.
>>>
>>> Thanks.
>>> Goran
>>>
>>>
>
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From mayamatakeshi at gmail.com  Thu Mar  4 15:55:35 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 5 Mar 2010 08:55:35 +0900
Subject: [Freeswitch-users] How to play back a .wav file from Erlang?
In-Reply-To: <4B8FEC3F.5060102@marketelsystems.com>
References: <4B8FEC3F.5060102@marketelsystems.com>
Message-ID: <15b9404e1003041555u12978339tdc35eb6d554fe341@mail.gmail.com>

On Fri, Mar 5, 2010 at 2:22 AM, Mark Sobkow <m.sobkow at marketelsystems.com>wrote:

> I have the UUID of a parked call.  I've waited n seconds for an operator
> to answer that call, but none are available so I want to play a message
> to the customer.  How do I do that from Erlang?  I've tried the
> following, but it doesn't seem to actually play the file, and I'm not
> getting any useful information in fs_cli about the playback attempt that
> would indicate why it's not working.
>
> bgapi( eval, "uuid:1f6d7a61-fd1a-4efd-bf84-31c4eea652c2 playback
> /opt/freeswitch/sounds/en/us/callie/misc/8000/sorry.wav" )
>

Command playback is used in the dialplan.
For parked calls you must use uuid_broadcast:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
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From brian at microcomaustralia.com.au  Thu Mar  4 16:00:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 5 Mar 2010 11:00:30 +1100
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B903D36.2040407@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
Message-ID: <3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>

On 5 March 2010 10:07, Yossi Neiman <freeswitch at cartissolutions.com> wrote:
> Standard response: ?It's in the oven baking right now. ?The muffin man
> will take it out as soon as it's ready...

Please ensure it is taken out of the oven in time and doesn't burn or
get overcooked.
-- 
Brian May <brian at microcomaustralia.com.au>


From jerry.richards at teotech.com  Thu Mar  4 16:01:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 16:01:30 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
Message-ID: <79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>

What I see in the log where they first differ is just after FS sends the
CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
didn't work.  Do you know what PICKED_OFF means?
 
In the good case, FS logs:
 
tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities

In the bad case, FS logs:
 
tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
PICKED_OFF
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
<mailto:sofia/internal/1059 at 192.168.72.141:5060>
sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
<mailto:sofia/internal/5402 at 192.168.72.141:5060>
sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]

Best Regards,
Jerry


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Thursday, March 04, 2010 2:59 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialing **<extension>




On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry



I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC



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From anthony.minessale at gmail.com  Thu Mar  4 16:14:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 18:14:18 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
	<79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
Message-ID: <191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>

Did you totally ignore my instructions?
Do you want help or not?


On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  What I see in the log where they first differ is just after FS sends the
> CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
> didn't work.  Do you know what PICKED_OFF means?
>
> In the good case, FS logs:
>
> tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060(already 1)
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
> ORIGINATOR_CANCEL
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
> sleep
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
> entities
> In the bad case, FS logs:
>
> tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060(already 1)
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
> CS_REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
> PICKED_OFF
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
> sleep
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
> entities
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
> sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
> [ORIGINATOR_CANCEL]
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Thursday, March 04, 2010 2:59 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>
>
>
> On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> By the way, it appears that the "**<extension>" ringing call pickup
>> feature
>> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
>> of
>> the following scenario:
>>
>> 1) 1059 calls 5381
>> 2) 5381 rings
>> 3) 5402 dials "**5381"
>>
>> In the bad case (http://pastebin.freeswitch.org/12325), all calls
>> disconnect
>> following above scenario.  In the good case
>> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>>
>> Best Regards,
>> Jerry
>>
>
> I'd suggest turning off the sofia debug stuff cuz that's just line noise
> unless you know all that stuff. You need to compare the two traces and look
> for differences. I'd start by looking at only the fs console output of each
> call and see if there's anything different between the two. See if there's a
> point where they diverge and work your way back from there. Look at the
> corresponding SIP dialogs and see if there are any clues as well. You may
> also want to capture the SIP traffic with tcpdump and analyze it in
> Wireshark which is easier on the eyes than having it buried in with the
> sofia and console logs.
>
> FWIW, I just updated to latest and I can't make this feature *NOT* work.
> I'm on 32-bit CentOS using Polycom 550 and Snom 320 phones.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Mar  4 16:19:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 18:19:13 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
	<3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>
Message-ID: <CD312DA2-4D85-47F0-8DCA-D0DB85453AD4@freeswitch.org>

?Find something you're passionate about and keep tremendously interested in it.?

/b

On Mar 4, 2010, at 6:00 PM, Brian May wrote:

> Please ensure it is taken out of the oven in time and doesn't burn or
> get overcooked.



From paul at apcl.us  Thu Mar  4 17:01:14 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 04 Mar 2010 20:01:14 -0500
Subject: [Freeswitch-users] how to see registration info for a
 particular user?
In-Reply-To: <4B902983.9080207@xpirio.com>
References: <4B8FCDA5.6000700@apcl.us> <4B902983.9080207@xpirio.com>
Message-ID: <4B9057DA.2010602@apcl.us>

Thanks Christian.  I thought I had tried that, but I guess I didn't.
It works.  Thanks!
     Paul


On 3/4/2010 4:43 PM, Christian L?schenkohl wrote:
> hi
>
> simply add the user id to the end of the command
> like "sofia status profile internal reg 123456"
> where 123456 is the id of the user you want to see
>
> br
>
> Paul Levin wrote:
>
>    
>> I am familiar with the console command:
>>       sofia status profile internal reg
>> to see the registration info for all currently registered users.
>>
>> Is there a modification to this command that will limit the result to
>> just one specific user registered user?
>>
>>     Thanks,
>>     Paul
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
>    



From jcasale at activenetwerx.com  Thu Mar  4 17:09:55 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 5 Mar 2010 01:09:55 +0000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>

>check this out ...?
>http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2
>
>I think?Goran Donev is Mr "Are we there yet" :)

Lol, you guys are just fscking merciless:)

Oh and btw, How 'bout now;)

/me Ducks...


From brian at freeswitch.org  Thu Mar  4 17:14:24 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:14:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>
Message-ID: <9CA09109-0FA1-4F29-A1AB-1EAB0B152EED@freeswitch.org>

How about NO.  We aren't there yet... got new batteries?

/b

On Mar 4, 2010, at 7:09 PM, Joseph L. Casale wrote:

> Oh and btw, How 'bout now;)



From infos at madovsky.org  Thu Mar  4 17:15:43 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 20:15:43 -0500
Subject: [Freeswitch-users] Release of 1.5
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
Message-ID: <B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>

Hey, this is my friend ! don't touch him !!
or I upgrade your faces with angry dog !! :D
  ----- Original Message ----- 
  From: jay binks 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 6:54 PM
  Subject: Re: [Freeswitch-users] Release of 1.5


  check this out ... 
  http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2


  I think Goran Donev is Mr "Are we there yet" :)


  J


  On Fri, Mar 5, 2010 at 9:30 AM, Yossi Neiman <freeswitch at cartissolutions.com> wrote:


    On 03/04/2010 05:15 PM, Anthony Minessale wrote: 
      You know, that's the only question you have asked on this list.

      I went back and searched for your name.

      And you have asked it like 12 times and have been answered the same each time.

      What is the obsession with an arbitrary tag?



    Maybe we should name the 1.0.5 official release as the "Donev Release"...






      On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

        No we have no deadline, we have spent countless hours preparing and we still have a small list of packaging concerns.  Feel free to download the code from http://latest.freeswitch.org and rest assured it *is* 1.0.5  



        On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <gorand at donevtechconsulting.com> wrote:

          Howdy Freeswitch community;

          I was wondering if the developers are any closer to the release of 1.05
          version. Please don't point me to an e-mail link :) Just inquiring as to
          whether there is a firm close date of 1.05.

          Thanks.
          Goran







Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com



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





  -- 
  Sincerely

  Jay



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Thu Mar  4 17:22:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:22:46 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
Message-ID: <8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>

I raise our angry dog... and see you a dirty dog...

/b

On Mar 4, 2010, at 7:15 PM, Madovsky wrote:

> Hey, this is my friend ! don't touch him !!
> or I upgrade your faces with angry dog !! :D
> ----- Original Message -----
> From: jay binks
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, March 04, 2010 6:54 PM
> Subject: Re: [Freeswitch-users] Release of 1.5
> 
> check this out ... 
> http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2
> 
> I think Goran Donev is Mr "Are we there yet" :)
> 
> J

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From lfurrea at gmail.com  Thu Mar  4 17:29:46 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 4 Mar 2010 19:29:46 -0600
Subject: [Freeswitch-users] event_socket outbound,
	trying to find out if a call gets put on hold 	by the phone
Message-ID: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>

Hi guys,

I am fooling (for the moment) around with event socket outbound and
basically I want to control a call that is made to a particular extension
and find out if this extension puts the call on hold. At the moment I don't
even know if the hold part is possible but I am having a hard time bridging
the call to a user.

Currently what I do is to direct the call made to user/301 to an outbound
socket, then I answer the channel and park it. Then I originate a channel to
user/301 and park it and then I try to use uuid_bridge to bridge both
channels and it seems that both channels drop.

Am I doing it correctly?

I am sure there are better ways to do it and I wanted to get some input from
you guys.

Thx a lot
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From gorand at donevtechconsulting.com  Thu Mar  4 17:34:55 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 19:34:55 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 48
In-Reply-To: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0d1e01cabc04$10896010$319c2030$@com>

I am having problems with my production server and I can not mess with
interim builds not knowing what else they can break.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
freeswitch-users-request at lists.freeswitch.org
Sent: Thursday, March 04, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 45, Issue 48

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of FreeSWITCH-users digest..."



From Russell.Mosemann at cune.org  Thu Mar  4 17:36:02 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Thu, 4 Mar 2010 19:36:02 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com><191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com><4B90428F.1000408@cartissolutions.com><ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com><B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
Message-ID: <AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>

Brian West wrote:
> I raise our angry dog... and see you a dirty dog...

OK, everyone just needs to kick back and enjoy a Mad Dog. :-)

--
Russell Mosemann



From brian at freeswitch.org  Thu Mar  4 17:42:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:42:17 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com><191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com><4B90428F.1000408@cartissolutions.com><ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com><B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
Message-ID: <33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>

I think you mean redbull.

/b

On Mar 4, 2010, at 7:36 PM, Russell Mosemann wrote:

> OK, everyone just needs to kick back and enjoy a Mad Dog. :-)



From rupa at rupa.com  Thu Mar  4 17:43:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Mar 2010 19:43:13 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 48
In-Reply-To: <0d1e01cabc04$10896010$319c2030$@com>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
	<0d1e01cabc04$10896010$319c2030$@com>
Message-ID: <b39273531003041743s227d677fl835c3cc9787b514@mail.gmail.com>

On Thu, Mar 4, 2010 at 7:34 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> I am having problems with my production server and I can not mess with
> interim builds not knowing what else they can break.
>

If you are going to participate in the mailing list, please switch to
non-digest mode.

Current trunk is more stable than whatever old version you are currently
running....  Try it out in a lab machine and after verifying go for it.  No
need to wait for 1.0.5.

-- 
-Rupa
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From infos at madovsky.org  Thu Mar  4 19:55:22 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 22:55:22 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
Message-ID: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>

Is there any example of how to use pgsql as db core on wiki ?
I tried to adapt this example
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
for pgsql but no success.
I'm trying to connect to it fhru openvpn.
tcpdump -i tap1 on the destination server doesn't show any packet exchange.
I set odbc.ini as this :

[freeswitch-pgsql]
Driver   = /usr/lib64/psqlodbc.so
SERVER   = 10.0.0.1
PORT     = 5432
DATABASE = freeswitch

and set all xml conf where odbc-dsn is as this :
<param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

didn't find anything interesting on google

Thanks

Franck
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From infos at madovsky.org  Thu Mar  4 19:59:03 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 22:59:03 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
Message-ID: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>

forgot debug :

2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
Could not connect to remote socket immedaitely

2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

thx

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 10:55 PM
  Subject: freeswitch and postgresql as db core


  Is there any example of how to use pgsql as db core on wiki ?
  I tried to adapt this example
  http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
  for pgsql but no success.
  I'm trying to connect to it fhru openvpn.
  tcpdump -i tap1 on the destination server doesn't show any packet exchange.
  I set odbc.ini as this :

  [freeswitch-pgsql]
  Driver   = /usr/lib64/psqlodbc.so
  SERVER   = 10.0.0.1
  PORT     = 5432
  DATABASE = freeswitch

  and set all xml conf where odbc-dsn is as this :
  <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

  didn't find anything interesting on google

  Thanks

  Franck
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From lakindia89 at gmail.com  Thu Mar  4 20:06:16 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 5 Mar 2010 09:36:16 +0530
Subject: [Freeswitch-users] event_socket outbound,
	trying to find out if a 	call gets put on hold by the phone
In-Reply-To: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>
References: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>
Message-ID: <7d79b3931003042006v1720f8a8g7a0eb0b3728e1dc2@mail.gmail.com>

Can you please pastebin the dialplan! as well as the error!


On Fri, Mar 5, 2010 at 6:59 AM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi guys,
>
> I am fooling (for the moment) around with event socket outbound and
> basically I want to control a call that is made to a particular extension
> and find out if this extension puts the call on hold. At the moment I don't
> even know if the hold part is possible but I am having a hard time bridging
> the call to a user.
>
> Currently what I do is to direct the call made to user/301 to an outbound
> socket, then I answer the channel and park it. Then I originate a channel to
> user/301 and park it and then I try to use uuid_bridge to bridge both
> channels and it seems that both channels drop.
>
> Am I doing it correctly?
>
> I am sure there are better ways to do it and I wanted to get some input
> from you guys.
>
> Thx a lot
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Mar  4 20:11:18 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 5 Mar 2010 09:41:18 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
Message-ID: <7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>

Ok. This is to say how the problem got solved.
Need openzap 1047 or above version.
Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz
My telco is not accepting the display IE.
Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.

Thanks for all.

On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn
<tomek.augustyn at gmail.com>wrote:

> Hello,
>
> I had similar problem and I think it is more a problem between Sangoma card
> and your E1 provider than with freeswitch.
>
> In my case it was necessary to set "origination_caller_id_number" to one
> of the telephone numbers linked to my E1 line. In other case the calls were
> rejected with "invalid information element" error.
>
> You can try Sangoma's support they are very helpful.
>
> Tomasz Augustyn
>
>
> ---------- Forwarded message ----------
> From: lakshmanan ganapathy <lakindia89 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 27 Feb 2010 10:32:08 +0530
> Subject: Re: [Freeswitch-users] smg_prid not bridging the call
> In the Dchan log it is saying Invalid Information Elements. That might be a
> problem??? But I even don't know why it is saying Invalid Information
> Element??
> Please guide me!!!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Mar  4 21:00:20 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 00:00:20 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core [SOLVED]
Message-ID: <489E6063C33B46119F9C7B4DE9E9FB50@MOBILEE1705>

to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

odbcinst.ini

[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib64/psqlodbc.so
Setup           = /usr/lib64/libodbcpsqlS.so
FileUsage       = 1

odbc.ini

[ODBC Data Sources]
freeswitch-pgsql = ODBC for PostgreSQL

[freeswitch-pgsql]
Description = ODBC for PostgreSQL
Driver   = /usr/lib64/psqlodbc.so
Servername   = 10.0.0.1
UserName = login
Password = pwd
Port     = 5432
Database = dbdb


Thanks me ;D
  ----- Original Message ----- 
  From: Madovsky 
  To: Madovsky 
  Sent: Thursday, March 04, 2010 10:58 PM
  Subject: Re: freeswitch and postgresql as db core


  forgot debug :

  2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
  Could not connect to remote socket immedaitely

  2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

  thx

  F


    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 04, 2010 10:55 PM
    Subject: freeswitch and postgresql as db core


    Is there any example of how to use pgsql as db core on wiki ?
    I tried to adapt this example
    http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
    for pgsql but no success.
    I'm trying to connect to it fhru openvpn.
    tcpdump -i tap1 on the destination server doesn't show any packet exchange.
    I set odbc.ini as this :

    [freeswitch-pgsql]
    Driver   = /usr/lib64/psqlodbc.so
    SERVER   = 10.0.0.1
    PORT     = 5432
    DATABASE = freeswitch

    and set all xml conf where odbc-dsn is as this :
    <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

    didn't find anything interesting on google

    Thanks

    Franck
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From craig at overthewire.com.au  Thu Mar  4 21:01:43 2010
From: craig at overthewire.com.au (Craig Askings)
Date: Fri, 5 Mar 2010 15:01:43 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
	<33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
Message-ID: <8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>

Bah, Jolt is the one true caffeinated beverage.

Craig.

On 5 March 2010 11:42, Brian West <brian at freeswitch.org> wrote:

> I think you mean redbull.
>
> /b
>
> On Mar 4, 2010, at 7:36 PM, Russell Mosemann wrote:
>
> > OK, everyone just needs to kick back and enjoy a Mad Dog. :-)
>
> --
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From nagalenoj at gmail.com  Thu Mar  4 21:07:10 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 5 Mar 2010 10:37:10 +0530
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
Message-ID: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>

Dear friends,
     I've faced an issue in event socket. I would want to know why it
behaves such a way.

     My program is working with the hep of events. So, based on the received
event, the process will continue it's work. When I need DTMF, I use
recvEventTimed and in the other cases, I use recvEvent.
     So, In the mid if caller hangsup, I expect for the SERVER_DISCONNECTED
event. When the caller hangsup when I'm waiting in recvEvent, I'm getting
SERVER_DISCONNECTED. But, when I'm waiting in recvEventTimed, I'mnot
receiving SERVER_DISCONNECTED, instead receiving an undefined value.

    To handle this, I've checked esl connection inside the timeout part and
put a recvEvent, then I receive SERVER_DISCONNECTED.

    My question is, why am I not receiving SERVER_DISCONNECTED when I recv
event using recvEventTimed?!

This is a sample program I execute,

require ESL;
use IO::Socket::INET;

my $ip = "127.0.0.1";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '9242',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;

for(;;) {
  my $new_sock = $sock->accept();
  my $pid = fork();
  if ($pid) {
    close($new_sock);
    next;
  }

  my $host = $new_sock->sockhost();
  my $fd = fileno($new_sock);

  my $con = new ESL::ESLconnection($fd);
  my $info = $con->getInfo();
  my $uuid = $info->getHeader("unique-id");
  printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");

  my $e = $con->filter("unique-id", $uuid);
  if ($e) {
    print $e->serialize();
  } else {
    printf("WTF?\n");
  }

  $con->events("plain", "SERVER_DISCONNECTED DTMF");
  $con->execute("answer");

  while($con->connected()) {
    $e = $con->recvEvent();   ############# CASE 1 ##
    # $e = $con->recvEventTimed(10000);       ############# CASE 2 ##

    unless ($e) {
        unless ($con->connected()) {
###############################
           $e = $con->recvEvent();                         ##
           print $e->serialize();                               ##   Added
code to get SERVER_DISCONNECTED.
           print "SERVER_DISCONNECTED";     ##
        }
###############################
        print "DTMF timeout\n";
    }
    if ($e) {
      my $name = $e->getHeader("event-name");
      print "EVENT [$name]\n";
      if ($name eq "DTMF") {
        my $digit = $e->getHeader("dtmf-digit");
        my $duration = $e->getHeader("dtmf-duration");
        print "DTMF digit $digit ($duration)\n";
      }
    }
  }
  print "BYE\n";
  close($new_sock);
}

-- 
Regards,
Nagalenoj H.
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From freeswitch.org at todandlorna.com  Thu Mar  4 21:25:22 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Thu, 04 Mar 2010 22:25:22 -0700
Subject: [Freeswitch-users] Changing Title just for Goran!
In-Reply-To: <0d1e01cabc04$10896010$319c2030$@com>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
	<0d1e01cabc04$10896010$319c2030$@com>
Message-ID: <4B9095C2.8080508@todandlorna.com>

Goran,

I can understand your desire for production stable versions.  You are 
obviously looking for stability for your client(s) and any risk you put 
them in you incur.  I have similar concerns in my own work, where 
updating FreeSWITCH could potentially cause issues that might cost the 
company money for downtime and headaches for me.  I empathize.

For a moment, please return this courtesy and try to understand this 
from the developers' perspective.  I believe you have (if you don't 
currently) done tech support for others.  Nagging didn't help you solve 
the problem any faster and often time, it makes the problem more 
difficult to approach with such a distraction.  This is effectively what 
you are doing for your needs.

Exercise some patience, or for a more timely solution, realize something 
about the FreeSWITCH project.  This is not a company with a Q-A 
department or a robust test-lab.  This is a group of individuals working 
on open-source software in various locations with various testing 
processes in place, and further help needed.  Thus, their current 
"trunk" from SVN has to be stable for their multitude of test processes, 
and code review has more relevance than Q-A.

Try trunk.  It will be just as stable for 90% of uses as 1.0.5 release 
will be.  Do a proper backup and a test run.  If you see problems, roll 
back.  It's not hard, it gets you resolved fast, and the project doesn't 
get derailed by constant questions about when it can get done.

Perhaps you'd like to give back to the project your client is relying 
on.  In that case, you can do some Q-A and test as well.  Work in Jira 
if you have such skills and do the testing.  If not, that's ok too. 
Just let other capable hands do it on their own timetable.  They're not 
lazy, and they're not unaware of the demand for 1.0.5.  It would do 
wonders if you could acknowledge that talent working for your benefit 
instead of conveying your displeasure that it is not happening on your 
specific timetable.

Cheers,

-Tod Hansmann

On 3/4/2010 6:34 PM, Goran Donev wrote:
> I am having problems with my production server and I can not mess with
> interim builds not knowing what else they can break.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> freeswitch-users-request at lists.freeswitch.org
> Sent: Thursday, March 04, 2010 5:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: FreeSWITCH-users Digest, Vol 45, Issue 48
>
> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of FreeSWITCH-users digest..."
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Fri Mar  5 00:20:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:20:02 -0500
Subject: [Freeswitch-users] TEST
In-Reply-To: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>
References: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>
Message-ID: <F07A3EC7-36AB-4C5D-A965-976F8BA902D5@jerris.com>

sure, your email provider now flagged everything from this address or ip as spam, this is of course totally normal.

On Mar 4, 2010, at 8:57 AM, Milena wrote:

> you failed!!  O_O
> after the messages from clubWNC, I'm not getting any more list messages on my e-mail inbox, is it normal?



From mike at jerris.com  Fri Mar  5 00:31:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:31:02 -0500
Subject: [Freeswitch-users] about pizza demo
In-Reply-To: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
References: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
Message-ID: <1F4E6321-3C01-4994-83A0-ABE4D6BCDC75@jerris.com>

no, it would require much much more, first step is to understand how your ASR engine deals with other languages.  This is a complicated task well outside the expertise of this list and best suited to the support resources for the specific asr engine.

Mike

On Feb 26, 2010, at 5:39 AM, Madovsky wrote:

> Hi,
>  
> I'm trying to change the language of pizza demo script.
> is it need to change only words inside addItemAlias() ?
>  

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From mike at jerris.com  Fri Mar  5 00:34:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:34:35 -0500
Subject: [Freeswitch-users] Video pass problem
In-Reply-To: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
References: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
Message-ID: <FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>

double check if the behavior on this is the same with svn trunk of FreeSWITCH on something other than a bleeding edge operating system.

Mike

On Feb 25, 2010, at 7:59 AM, Mikhail Krivushin wrote:

> Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and see that we have 1280 video packets input, and only 560 passed to B leg. Anyone can point me to right direction? I can send pcap file by request.
> 
> Most time I have black screen, and then one frame can appear, and I see frozen picture pair minutes, and then other frozen picture. We have not network issues, we have good video, when bypass FS. We have not perfomance troubles to.
> 
> We have ubuntu 9.10 x64, and powerfull server board. uname:
> Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux
> 
> config:
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
> 
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From tculjaga at gmail.com  Fri Mar  5 01:44:55 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Mar 2010 10:44:55 +0100
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
	<ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
Message-ID: <65d96fc81003050144k76ffdf70rea30b109e2b19392@mail.gmail.com>

On Thu, Mar 4, 2010 at 11:25 PM, jay binks <jaybinks at gmail.com> wrote:

> HUH, well there you go..   exactly what I was after...
> someone has done exactly what I was thinking.
>
> I know you've sent it to me by email, but lets get this in the wiki.
> Ill put this up from your email, but if you have more id encourage you to
> share what youve done.
>
>

let me finish the current project ..."wholesale routing machine". and i will
put everything on wiki



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

Hi,

 

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

 

Tried the scenario as follows:

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

 

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

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

 

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

 

Thanks & Regards

Suneel


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From ahmed.ajmal at breezecom.ae  Fri Mar  5 05:02:46 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Fri, 5 Mar 2010 18:02:46 +0500
Subject: [Freeswitch-users] Losing Log lines
Message-ID: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>

Hi 

 

I am seeing the following "CRIT" log lines in my freeswitch logs:

 

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:49.633560 [CRIT]
mod_event_socket.c:161 Lost 134 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.374504 [CRIT]
mod_event_socket.c:161 Lost 218 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.893504 [CRIT]
mod_event_socket.c:161 Lost 143 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:59.093785 [CRIT]
mod_event_socket.c:161 Lost 27 log lines!

 

And this is happening too often now. I am relying on ESL to capture
HANGUP_COMPLETE event and insert the cdr fields in a database and it appears
from the above lines that freeswitch is missing some logs here. I am running
freeswitch on a 64-bit version of CentOS 5.4 with 4 gigs of RAM. There were
around 72 active calls(fs_cli -x "show calls count") when I noticed the
above. Also heres what my max sessions look like:

 

<param name="max-sessions" value="4000"/>

    <!--Most channels to create per second -->

<param name="sessions-per-second" value="100"/>

 

I am not sure if this is a memory issue or not but I also noticed that the
machine was running on very low memory when all this was happening. Heres
what my free -m stats looked like:

 

                       total       used       free     shared    buffers
cached

Mem:          3952       3926         26          0        186       1156

-/+ buffers/cache:        264       3688

Swap:          8189          0       8189

 

Please help me with any resolution and/or suggestions.

 

Thanks

Ahmed

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From abrownworth at bandwidth.com  Fri Mar  5 06:29:02 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Fri, 5 Mar 2010 09:29:02 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
In-Reply-To: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>
References: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>
Message-ID: <dbad22fd1003050629u772958eds6efa6e36385631a1@mail.gmail.com>

Try telnetting from your FreeSWITCH box to 10.0.0.1:

shell#> telnet 10.0.0.1 5432

if it doesn't connect, then this isn't a FreeSWITCH / unixODBC error. Check
your routing, etc.

If it does connect, then the error looks unixODBC related.

-a

On Thu, Mar 4, 2010 at 10:59 PM, Madovsky <infos at madovsky.org> wrote:

>  forgot debug :
>
> 2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101
> ERROR: [unixODBC]Could not connect to the server;
> Could not connect to remote socket immedaitely
>
> 2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to
> ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!
>
> thx
>
> F
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 04, 2010 10:55 PM
> *Subject:* freeswitch and postgresql as db core
>
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver   = /usr/lib64/psqlodbc.so
> SERVER   = 10.0.0.1
> PORT     = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Mar  5 06:32:55 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 08:32:55 -0600
Subject: [Freeswitch-users] Losing Log lines
In-Reply-To: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>
References: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>
Message-ID: <A4B7DF27-79F3-4DB2-88DE-A19BAE249A31@freeswitch.org>

You're not reading everything off the socket so it has to drop them at some point.

/b

On Mar 5, 2010, at 7:02 AM, Ahmed Ajmal wrote:

> Hi
>  
> I am seeing the following ?CRIT? log lines in my freeswitch logs:
>  
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:49.633560 [CRIT] mod_event_socket.c:161 Lost 134 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.374504 [CRIT] mod_event_socket.c:161 Lost 218 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.893504 [CRIT] mod_event_socket.c:161 Lost 143 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:59.093785 [CRIT] mod_event_socket.c:161 Lost 27 log lines!
>  
> And this is happening too often now. I am relying on ESL to capture HANGUP_COMPLETE event and insert the cdr fields in a database and it appears from the above lines that freeswitch is missing some logs here. I am running freeswitch on a 64-bit version of CentOS 5.4 with 4 gigs of RAM. There were around 72 active calls(fs_cli -x "show calls count") when I noticed the above. Also heres what my max sessions look like:
>  
> <param name="max-sessions" value="4000"/>
>     <!--Most channels to create per second -->
> <param name="sessions-per-second" value="100"/>
>  
> I am not sure if this is a memory issue or not but I also noticed that the machine was running on very low memory when all this was happening. Heres what my free -m stats looked like:
>  
>                        total       used       free     shared    buffers     cached
> Mem:          3952       3926         26          0        186       1156
> -/+ buffers/cache:        264       3688
> Swap:          8189          0       8189
>  
> Please help me with any resolution and/or suggestions.
>  
> Thanks
> Ahmed
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dome at tel.co.th  Fri Mar  5 08:00:01 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 5 Mar 2010 23:00:01 +0700
Subject: [Freeswitch-users] freeswitch and postgresql as db core
In-Reply-To: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>
References: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>
Message-ID: <8ccbff061003050800h12f720e9uba448422065b92fb@mail.gmail.com>

I just update wiki
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#Postgresql_8.4_on_Debian_Squeeze
It's work fine forme.

Dome C.


2010/3/5 Madovsky <infos at madovsky.org>:
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver?? = /usr/lib64/psqlodbc.so
> SERVER?? = 10.0.0.1
> PORT???? = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From m.sobkow at marketelsystems.com  Fri Mar  5 08:42:39 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 05 Mar 2010 10:42:39 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
Message-ID: <4B91347F.6070509@marketelsystems.com>


>>>         
>>     Try using park_after_bridge and catch the park event or
>>     transfer_after_bridge and set the dialplan location to transfer the call
>>     to.
>>
>>     Andrew
>>
>>       
>     What a relief.  "transfer_after_bridge" worked!  Plus it provided
>     an obvious way for me to collect the call result code, fire it
>     back to Erlang, and leave the operator waiting for the next
>     incoming call. :)
>
> "Another satisfied customer!" :D Now it's time to throw some of that 
> Canadian colored money at Tony's paypal account. ;)
> BTW, what kind of solution are you developing?
> -MC
Tech support call center solution right now, with bidirectional 
calling.  If a customer calls in and no one is available, they have the 
option of requesting a callback instead of waiting online, so both 
inbound and outbound calls have to be handled.  I'm also having fun with 
skill-based routing of calls (which is more complex than mod_fifo can 
handle.)


From infos at madovsky.org  Fri Mar  5 08:45:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 11:45:04 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db core [SOLVED]
Message-ID: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>

I sent An email last night but didn't work,
so I retry it now...


to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

odbcinst.ini

[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib64/psqlodbc.so
Setup           = /usr/lib64/libodbcpsqlS.so
FileUsage       = 1

odbc.ini

[ODBC Data Sources]
freeswitch-pgsql = ODBC for PostgreSQL

[freeswitch-pgsql]
Description = ODBC for PostgreSQL
Driver   = /usr/lib64/psqlodbc.so
Servername   = 10.0.0.1
UserName = login
Password = pwd
Port     = 5432
Database = dbdb


Thanks me ;D
  ----- Original Message ----- 
  From: Madovsky 
  To: Madovsky 
  Sent: Thursday, March 04, 2010 10:58 PM
  Subject: Re: freeswitch and postgresql as db core


  forgot debug :

  2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
  Could not connect to remote socket immedaitely

  2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

  thx

  F


    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 04, 2010 10:55 PM
    Subject: freeswitch and postgresql as db core


    Is there any example of how to use pgsql as db core on wiki ?
    I tried to adapt this example
    http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
    for pgsql but no success.
    I'm trying to connect to it fhru openvpn.
    tcpdump -i tap1 on the destination server doesn't show any packet exchange.
    I set odbc.ini as this :

    [freeswitch-pgsql]
    Driver   = /usr/lib64/psqlodbc.so
    SERVER   = 10.0.0.1
    PORT     = 5432
    DATABASE = freeswitch

    and set all xml conf where odbc-dsn is as this :
    <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

    didn't find anything interesting on google

    Thanks

    Franck
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From anthony.minessale at gmail.com  Fri Mar  5 08:45:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 10:45:34 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
	<33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
	<8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>
Message-ID: <191c3a031003050845u1fb2380bg6d0638379d2d29ae@mail.gmail.com>

No mad dog is the preferred beverage of winos across the land!

On Mar 4, 2010 11:07 PM, "Craig Askings" <craig at overthewire.com.au> wrote:

Bah, Jolt is the one true caffeinated beverage.

Craig.



On 5 March 2010 11:42, Brian West <brian at freeswitch.org> wrote:
>
> I think you mean redbull.
>
> ...
-- 
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From m.sobkow at marketelsystems.com  Fri Mar  5 08:45:42 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 05 Mar 2010 10:45:42 -0600
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
Message-ID: <4B913536.70407@marketelsystems.com>

Michael Collins wrote:
>
>
> On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com 
> <mailto:phunk0000 at hotmail.com>> wrote:
>
>     Hey List..  I am trying to setup my dial plan so whenever a call
>     is answered a recording is played to the person who answered the
>     phone before the call is connected?.  Any help would be great.
>       Thanks
>
> Are you talking about the group confirm feature?
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> -MC
You might also try uuid_broadcast if you're doing code-driven 
dialplans.  That's what I ended up using for my Erlang-Freeswitch 
integration.


From infos at madovsky.org  Fri Mar  5 08:53:55 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 11:53:55 -0500
Subject: [Freeswitch-users] about pizza demo
References: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
	<1F4E6321-3C01-4994-83A0-ABE4D6BCDC75@jerris.com>
Message-ID: <796886F978564D60A19E8B282E810127@MOBILEE1705>

Mike,

I took time to learn more about this feature,
I understand now how it works. If I have time
I will try to create an ASR engine for French and Italian

Regards

  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 3:31 AM
  Subject: Re: [Freeswitch-users] about pizza demo


  no, it would require much much more, first step is to understand how your ASR engine deals with other languages.  This is a complicated task well outside the expertise of this list and best suited to the support resources for the specific asr engine.


  Mike


  On Feb 26, 2010, at 5:39 AM, Madovsky wrote:


    Hi,

    I'm trying to change the language of pizza demo script.
    is it need to change only words inside addItemAlias() ?





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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From abrownworth at bandwidth.com  Fri Mar  5 08:54:22 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Fri, 5 Mar 2010 11:54:22 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db core
	[SOLVED]
In-Reply-To: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
Message-ID: <dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>

I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity
through your tap1 device. Does this work:

telnet 10.0.0.1 5432

-Anders

On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

>  I sent An email last night but didn't work,
> so I retry it now...
>
> to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10
> update rpm and Postgresql source)
>
> odbcinst.ini
>
> [PostgreSQL]
> Description     = ODBC for PostgreSQL
> Driver          = /usr/lib64/psqlodbc.so
> Setup           = /usr/lib64/libodbcpsqlS.so
> FileUsage       = 1
> odbc.ini
>
> [ODBC Data Sources]
> freeswitch-pgsql = ODBC for PostgreSQL
>
> [freeswitch-pgsql]
> Description = ODBC for PostgreSQL
> Driver   = /usr/lib64/psqlodbc.so
> Servername   = 10.0.0.1
> UserName = login
> Password = pwd
> Port     = 5432
> Database = dbdb
>
> Thanks me ;D
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* Madovsky <infos at madovsky.org>
> *Sent:* Thursday, March 04, 2010 10:58 PM
> *Subject:* Re: freeswitch and postgresql as db core
>
> forgot debug :
>
> 2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101
> ERROR: [unixODBC]Could not connect to the server;
> Could not connect to remote socket immedaitely
>
> 2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to
> ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!
>
> thx
>
> F
>
>  ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 04, 2010 10:55 PM
> *Subject:* freeswitch and postgresql as db core
>
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver   = /usr/lib64/psqlodbc.so
> SERVER   = 10.0.0.1
> PORT     = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri Mar  5 09:06:39 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 12:06:39 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db
	core[SOLVED]
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
	<dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>
Message-ID: <D0C806F2267A419A92D60E140DCD57BE@MOBILEE1705>

No, it works without problem
  ----- Original Message ----- 
  From: Anders Brownworth 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 11:54 AM
  Subject: Re: [Freeswitch-users] Fw: freeswitch and postgresql as db core[SOLVED]


  I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity through your tap1 device. Does this work:

  telnet 10.0.0.1 5432

  -Anders


  On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

    I sent An email last night but didn't work,
    so I retry it now...


    to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

    odbcinst.ini

    [PostgreSQL]
    Description     = ODBC for PostgreSQL
    Driver          = /usr/lib64/psqlodbc.so
    Setup           = /usr/lib64/libodbcpsqlS.so
    FileUsage       = 1

    odbc.ini

    [ODBC Data Sources]
    freeswitch-pgsql = ODBC for PostgreSQL

    [freeswitch-pgsql]
    Description = ODBC for PostgreSQL
    Driver   = /usr/lib64/psqlodbc.so
    Servername   = 10.0.0.1
    UserName = login
    Password = pwd
    Port     = 5432
    Database = dbdb


    Thanks me ;D
      ----- Original Message ----- 
      From: Madovsky 
      To: Madovsky 
      Sent: Thursday, March 04, 2010 10:58 PM
      Subject: Re: freeswitch and postgresql as db core


      forgot debug :

      2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
      Could not connect to remote socket immedaitely

      2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

      thx

      F


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 04, 2010 10:55 PM
        Subject: freeswitch and postgresql as db core


        Is there any example of how to use pgsql as db core on wiki ?
        I tried to adapt this example
        http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
        for pgsql but no success.
        I'm trying to connect to it fhru openvpn.
        tcpdump -i tap1 on the destination server doesn't show any packet exchange.
        I set odbc.ini as this :

        [freeswitch-pgsql]
        Driver   = /usr/lib64/psqlodbc.so
        SERVER   = 10.0.0.1
        PORT     = 5432
        DATABASE = freeswitch

        and set all xml conf where odbc-dsn is as this :
        <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

        didn't find anything interesting on google

        Thanks

        Franck

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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Fri Mar  5 09:07:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 12:07:04 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db
	core[SOLVED]
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
	<dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>
Message-ID: <6AA8ACBF2F5C45E68DFA552625F8FD29@MOBILEE1705>

I wrote below, it solved
  ----- Original Message ----- 
  From: Anders Brownworth 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 11:54 AM
  Subject: Re: [Freeswitch-users] Fw: freeswitch and postgresql as db core[SOLVED]


  I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity through your tap1 device. Does this work:

  telnet 10.0.0.1 5432

  -Anders


  On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

    I sent An email last night but didn't work,
    so I retry it now...


    to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

    odbcinst.ini

    [PostgreSQL]
    Description     = ODBC for PostgreSQL
    Driver          = /usr/lib64/psqlodbc.so
    Setup           = /usr/lib64/libodbcpsqlS.so
    FileUsage       = 1

    odbc.ini

    [ODBC Data Sources]
    freeswitch-pgsql = ODBC for PostgreSQL

    [freeswitch-pgsql]
    Description = ODBC for PostgreSQL
    Driver   = /usr/lib64/psqlodbc.so
    Servername   = 10.0.0.1
    UserName = login
    Password = pwd
    Port     = 5432
    Database = dbdb


    Thanks me ;D
      ----- Original Message ----- 
      From: Madovsky 
      To: Madovsky 
      Sent: Thursday, March 04, 2010 10:58 PM
      Subject: Re: freeswitch and postgresql as db core


      forgot debug :

      2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
      Could not connect to remote socket immedaitely

      2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

      thx

      F


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 04, 2010 10:55 PM
        Subject: freeswitch and postgresql as db core


        Is there any example of how to use pgsql as db core on wiki ?
        I tried to adapt this example
        http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
        for pgsql but no success.
        I'm trying to connect to it fhru openvpn.
        tcpdump -i tap1 on the destination server doesn't show any packet exchange.
        I set odbc.ini as this :

        [freeswitch-pgsql]
        Driver   = /usr/lib64/psqlodbc.so
        SERVER   = 10.0.0.1
        PORT     = 5432
        DATABASE = freeswitch

        and set all xml conf where odbc-dsn is as this :
        <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

        didn't find anything interesting on google

        Thanks

        Franck

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






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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dome at tel.co.th  Fri Mar  5 09:14:39 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 00:14:39 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912) Problem.
Message-ID: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>

Dear All,
        I found big odbc problem when i try to move to FS 1.0.5.
in same server i'm testing 1.0.4 and 1.0.5 and same config
1. config  limit.conf.xml use odbc (i use postgresql)
 <param name="odbc-dsn" value="tuxza:admin:xxx"/>
2. create dialplan in context public for test limit
  <condition field="destination_number" expression="^(1111)$">
       <action application="limit" data="$${domain} 0838833133 500"/>
      <action application="answer"/>
      <action application="sleep" data="60000"/>
</condition>
3. use sipp generate test call
   sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
xx.xx.xx.xx:5080

1.0.4 work fine i found 2 connection in postgresql server
but 1.0.5 use 1 connection for 1 calls. and get error when call over
100 calls. (postgresql maximum client connect is 100 )

So i'm not sure its' BUG or not ?


BG

Dome C.


From msc at freeswitch.org  Fri Mar  5 09:24:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Mar 2010 09:24:52 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B91347F.6070509@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<4B91347F.6070509@marketelsystems.com>
Message-ID: <87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>

> > "Another satisfied customer!" :D Now it's time to throw some of that
> > Canadian colored money at Tony's paypal account. ;)
> > BTW, what kind of solution are you developing?
> > -MC
> Tech support call center solution right now, with bidirectional
> calling.  If a customer calls in and no one is available, they have the
> option of requesting a callback instead of waiting online, so both
> inbound and outbound calls have to be handled.  I'm also having fun with
> skill-based routing of calls (which is more complex than mod_fifo can
> handle.)
>

Very interesting. Okay, keep us posted on how your setup is coming along.
You might also want to join the Wednesday morning conference calls and talk
to some of the other users as well as listen to some of the presentations.
Next week Math will be discussing mod_sofia in excruciating detail. :)

-MC
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From brian at freeswitch.org  Fri Mar  5 09:32:30 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 11:32:30 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<4B91347F.6070509@marketelsystems.com>
	<87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>
Message-ID: <F3130139-91DF-491A-ACAB-8C1976A49422@freeswitch.org>

Better clear your schedule for a few days for that talk :P

/b

On Mar 5, 2010, at 11:24 AM, Michael Collins wrote:

> 
> Very interesting. Okay, keep us posted on how your setup is coming along. You might also want to join the Wednesday morning conference calls and talk to some of the other users as well as listen to some of the presentations. Next week Math will be discussing mod_sofia in excruciating detail. :)
> 
> -MC



From edpimentl at gmail.com  Fri Mar  5 10:16:55 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 5 Mar 2010 13:16:55 -0500
Subject: [Freeswitch-users] smart-http
Message-ID: <9dc4a1671003051016l11955a62wb2dde816cf3b1d13@mail.gmail.com>

This is just an FYI and get your thinking on the value of smart-http for
future use.
http://progit.org/2010/03/04/smart-http.html

-E
http://vCardCloud.com
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From mike at van.lammeren.net  Fri Mar  5 11:04:54 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:04:54 -0500
Subject: [Freeswitch-users] Which Lua script am I?
Message-ID: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>

Hello!

I am working on a high-availability system and need a Lua script that can
run in the background and loop forever, periodically waking up to accomplish
some tasks. I would like to be able to handle at least 10 concurrent tasks.

My first idea was to have a simple background script that would call another
script, let's call it the 'task' script, every 5 seconds or so. This would
spread the work around, and if any task script quit with an error, the
background script would remain running, and would continue to launch new
task scripts.

However, I can see no way for the background script to launch other Lua
scripts. Am I missing something? I need something like session:execute(),
but unfortunately this background script has no session, and I'm not sure if
session:execute() can even make a call like the "luarun" console command.

So, my next ideas was to have 10 copies of the background script. Each
script would loop forever. Upon launch, each script would perform a single
task, then sleep for 60 seconds. This allows handling of 10 concurrent
tasks.

However, there are some problems with running 10 copies of the same script,
which I believe can be solved if each script could be given a unique
identifier. One problem is that I would like the scripts to actually sleep
for a variable amount of time, say between 55 and 65 seconds, so that the
workload would spread around. Another, similar problem, is that each script
needs to 'claim' a number of database records from a table that is queuing
up work for the scripts. Once again, if each script had a unique identifier,
then I could use an update query to effectively 'set and test' a record that
would be tied to that script instance.

I could use a random number to identify each script, but since math.random
is seeded with a timestamp that only resolves to seconds, then all 10
scripts will be given the same sequence of random numbers, and all 10 will
run in lock-step with each other. So, random numbers are out. (If I had a
unique ID to seed the random number generator, then I could just use that
unique ID and wouldn't need random numbers.)

Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
launches at startup?

Or is there a better way of handling this situation altogether? I'm open to
any suggestions that people would care to make!

(I would entertain the idea of a single script that would start with
FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
but that brings me back to the original problem of not knowing how to launch
other Lua scripts under FreeSWITCH.)

Mike van Lammeren
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From msc at freeswitch.org  Fri Mar  5 11:16:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Mar 2010 11:16:26 -0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>

May I ask what the end goal is? It could be that there are better tools in
the toolbox to help you...

-MC

On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I am working on a high-availability system and need a Lua script that can
> run in the background and loop forever, periodically waking up to accomplish
> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>
> My first idea was to have a simple background script that would call
> another script, let's call it the 'task' script, every 5 seconds or so. This
> would spread the work around, and if any task script quit with an error, the
> background script would remain running, and would continue to launch new
> task scripts.
>
> However, I can see no way for the background script to launch other Lua
> scripts. Am I missing something? I need something like session:execute(),
> but unfortunately this background script has no session, and I'm not sure if
> session:execute() can even make a call like the "luarun" console command.
>
> So, my next ideas was to have 10 copies of the background script. Each
> script would loop forever. Upon launch, each script would perform a single
> task, then sleep for 60 seconds. This allows handling of 10 concurrent
> tasks.
>
> However, there are some problems with running 10 copies of the same script,
> which I believe can be solved if each script could be given a unique
> identifier. One problem is that I would like the scripts to actually sleep
> for a variable amount of time, say between 55 and 65 seconds, so that the
> workload would spread around. Another, similar problem, is that each script
> needs to 'claim' a number of database records from a table that is queuing
> up work for the scripts. Once again, if each script had a unique identifier,
> then I could use an update query to effectively 'set and test' a record that
> would be tied to that script instance.
>
> I could use a random number to identify each script, but since math.random
> is seeded with a timestamp that only resolves to seconds, then all 10
> scripts will be given the same sequence of random numbers, and all 10 will
> run in lock-step with each other. So, random numbers are out. (If I had a
> unique ID to seed the random number generator, then I could just use that
> unique ID and wouldn't need random numbers.)
>
> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
> launches at startup?
>
> Or is there a better way of handling this situation altogether? I'm open to
> any suggestions that people would care to make!
>
> (I would entertain the idea of a single script that would start with
> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
> but that brings me back to the original problem of not knowing how to launch
> other Lua scripts under FreeSWITCH.)
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Mar  5 11:16:39 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 13:16:39 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <7C4D5C83-9A67-4024-BC68-4B92353545CD@freeswitch.org>

What exactly are your scripts doing?

/b

On Mar 5, 2010, at 1:04 PM, Mike van Lammeren wrote:

> Or is there a better way of handling this situation altogether? I'm open to any suggestions that people would care to make!



From mike at van.lammeren.net  Fri Mar  5 11:32:39 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:32:39 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
Message-ID: <5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>

I can only talk about the project in vague terms, due to client
confidentiality, but here goes:

Many thousands of devices will be reporting events, via HTTP Post, to a
central server farm. There will be at least 5 or 6 web servers recording
these events, and a similar number of VoIP servers running FreeSWITCH.
Communication between the web servers and VoIP servers is via a shared MySQL
Cluster. Certain events will result in FreeSWITCH originating a phone call,
playing a voice message to a human, then recording a number typed in by that
human.

It is important that each event result in just one phone call. We don't want
people bothered with multiple, simultaneous calls, and we don't want things
to be missed, either.

What do you think?


On Fri, Mar 5, 2010 at 2:16 PM, Michael Collins <msc at freeswitch.org> wrote:

> May I ask what the end goal is? It could be that there are better tools in
> the toolbox to help you...
>
> -MC
>
> On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> I am working on a high-availability system and need a Lua script that can
>> run in the background and loop forever, periodically waking up to accomplish
>> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>>
>> My first idea was to have a simple background script that would call
>> another script, let's call it the 'task' script, every 5 seconds or so. This
>> would spread the work around, and if any task script quit with an error, the
>> background script would remain running, and would continue to launch new
>> task scripts.
>>
>> However, I can see no way for the background script to launch other Lua
>> scripts. Am I missing something? I need something like session:execute(),
>> but unfortunately this background script has no session, and I'm not sure if
>> session:execute() can even make a call like the "luarun" console command.
>>
>> So, my next ideas was to have 10 copies of the background script. Each
>> script would loop forever. Upon launch, each script would perform a single
>> task, then sleep for 60 seconds. This allows handling of 10 concurrent
>> tasks.
>>
>> However, there are some problems with running 10 copies of the same
>> script, which I believe can be solved if each script could be given a unique
>> identifier. One problem is that I would like the scripts to actually sleep
>> for a variable amount of time, say between 55 and 65 seconds, so that the
>> workload would spread around. Another, similar problem, is that each script
>> needs to 'claim' a number of database records from a table that is queuing
>> up work for the scripts. Once again, if each script had a unique identifier,
>> then I could use an update query to effectively 'set and test' a record that
>> would be tied to that script instance.
>>
>> I could use a random number to identify each script, but since math.random
>> is seeded with a timestamp that only resolves to seconds, then all 10
>> scripts will be given the same sequence of random numbers, and all 10 will
>> run in lock-step with each other. So, random numbers are out. (If I had a
>> unique ID to seed the random number generator, then I could just use that
>> unique ID and wouldn't need random numbers.)
>>
>> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
>> launches at startup?
>>
>> Or is there a better way of handling this situation altogether? I'm open
>> to any suggestions that people would care to make!
>>
>> (I would entertain the idea of a single script that would start with
>> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
>> but that brings me back to the original problem of not knowing how to launch
>> other Lua scripts under FreeSWITCH.)
>>
>> Mike van Lammeren
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Mar  5 11:36:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 13:36:25 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <191c3a031003051136n36f6aed1m4adfdcd925ffaa71@mail.gmail.com>

indeed you are missing something.

The FSAPI interface, see the API object in the lua docs and the "luarun" api
command.

You will probably find this knowledge very useful but even more important
you may want to know that high-availability and script do not go together as
well as high-availability and C, for instance.


On Fri, Mar 5, 2010 at 1:04 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I am working on a high-availability system and need a Lua script that can
> run in the background and loop forever, periodically waking up to accomplish
> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>
> My first idea was to have a simple background script that would call
> another script, let's call it the 'task' script, every 5 seconds or so. This
> would spread the work around, and if any task script quit with an error, the
> background script would remain running, and would continue to launch new
> task scripts.
>
> However, I can see no way for the background script to launch other Lua
> scripts. Am I missing something? I need something like session:execute(),
> but unfortunately this background script has no session, and I'm not sure if
> session:execute() can even make a call like the "luarun" console command.
>
> So, my next ideas was to have 10 copies of the background script. Each
> script would loop forever. Upon launch, each script would perform a single
> task, then sleep for 60 seconds. This allows handling of 10 concurrent
> tasks.
>
> However, there are some problems with running 10 copies of the same script,
> which I believe can be solved if each script could be given a unique
> identifier. One problem is that I would like the scripts to actually sleep
> for a variable amount of time, say between 55 and 65 seconds, so that the
> workload would spread around. Another, similar problem, is that each script
> needs to 'claim' a number of database records from a table that is queuing
> up work for the scripts. Once again, if each script had a unique identifier,
> then I could use an update query to effectively 'set and test' a record that
> would be tied to that script instance.
>
> I could use a random number to identify each script, but since math.random
> is seeded with a timestamp that only resolves to seconds, then all 10
> scripts will be given the same sequence of random numbers, and all 10 will
> run in lock-step with each other. So, random numbers are out. (If I had a
> unique ID to seed the random number generator, then I could just use that
> unique ID and wouldn't need random numbers.)
>
> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
> launches at startup?
>
> Or is there a better way of handling this situation altogether? I'm open to
> any suggestions that people would care to make!
>
> (I would entertain the idea of a single script that would start with
> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
> but that brings me back to the original problem of not knowing how to launch
> other Lua scripts under FreeSWITCH.)
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Fri Mar  5 11:38:37 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:38:37 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
Message-ID: <5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>

So much for theory!

Take a look at this sample script.

-- math.randomseed(os.time())

for i=1,9 do -- final version will loop forever
    freeswitch.consoleLog("info", "wake-up #" .. i .. ", random number: " ..
math.random(111,999) .. "\n")
    -- Sleep for a while
    freeswitch.msleep(5000)
end

freeswitch.consoleLog("warning", "Script end.\n")

return 0


If I launch 10 copies of this Lua script with FreeSWITCH, with the
'startup-script' param in lua.conf.xml, then it turns out that each time
through the loop for each script does actually produce a unique random
number. If I comment out the first line, thus seeding the random number
generator, then all 10 scripts give the same 'random' number the first time
through the loop, then a unique random number for each subsequent loop!

Apparently, I don't really have the problem I thought I had.

All the same, I'm still open to suggestions on better ways to achieve my
ultimate goals!

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:32 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> I can only talk about the project in vague terms, due to client
> confidentiality, but here goes:
>
> Many thousands of devices will be reporting events, via HTTP Post, to a
> central server farm. There will be at least 5 or 6 web servers recording
> these events, and a similar number of VoIP servers running FreeSWITCH.
> Communication between the web servers and VoIP servers is via a shared MySQL
> Cluster. Certain events will result in FreeSWITCH originating a phone call,
> playing a voice message to a human, then recording a number typed in by that
> human.
>
> It is important that each event result in just one phone call. We don't
> want people bothered with multiple, simultaneous calls, and we don't want
> things to be missed, either.
>
> What do you think?
>
>
> On Fri, Mar 5, 2010 at 2:16 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> May I ask what the end goal is? It could be that there are better tools in
>> the toolbox to help you...
>>
>> -MC
>>
>> On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hello!
>>>
>>> I am working on a high-availability system and need a Lua script that can
>>> run in the background and loop forever, periodically waking up to accomplish
>>> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>>>
>>> My first idea was to have a simple background script that would call
>>> another script, let's call it the 'task' script, every 5 seconds or so. This
>>> would spread the work around, and if any task script quit with an error, the
>>> background script would remain running, and would continue to launch new
>>> task scripts.
>>>
>>> However, I can see no way for the background script to launch other Lua
>>> scripts. Am I missing something? I need something like session:execute(),
>>> but unfortunately this background script has no session, and I'm not sure if
>>> session:execute() can even make a call like the "luarun" console command.
>>>
>>> So, my next ideas was to have 10 copies of the background script. Each
>>> script would loop forever. Upon launch, each script would perform a single
>>> task, then sleep for 60 seconds. This allows handling of 10 concurrent
>>> tasks.
>>>
>>> However, there are some problems with running 10 copies of the same
>>> script, which I believe can be solved if each script could be given a unique
>>> identifier. One problem is that I would like the scripts to actually sleep
>>> for a variable amount of time, say between 55 and 65 seconds, so that the
>>> workload would spread around. Another, similar problem, is that each script
>>> needs to 'claim' a number of database records from a table that is queuing
>>> up work for the scripts. Once again, if each script had a unique identifier,
>>> then I could use an update query to effectively 'set and test' a record that
>>> would be tied to that script instance.
>>>
>>> I could use a random number to identify each script, but since
>>> math.random is seeded with a timestamp that only resolves to seconds, then
>>> all 10 scripts will be given the same sequence of random numbers, and all 10
>>> will run in lock-step with each other. So, random numbers are out. (If I had
>>> a unique ID to seed the random number generator, then I could just use that
>>> unique ID and wouldn't need random numbers.)
>>>
>>> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
>>> launches at startup?
>>>
>>> Or is there a better way of handling this situation altogether? I'm open
>>> to any suggestions that people would care to make!
>>>
>>> (I would entertain the idea of a single script that would start with
>>> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
>>> but that brings me back to the original problem of not knowing how to launch
>>> other Lua scripts under FreeSWITCH.)
>>>
>>> Mike van Lammeren
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From dave at 3c.co.uk  Fri Mar  5 11:40:14 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 5 Mar 2010 12:40:14 -0700
Subject: [Freeswitch-users] Which Lua script am I?
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <F14E9C432BF0489FB02210622CF267F9@DELL9>

Hi Mike,

You can spawn a new script with something like

api = freeswitch.API();
reply = api:executeString("luarun /tmp/msg.lua");

- and you can pass parameters, accessed in the normal way.

So there's no reason why you couldn't have a background script running which spawns one worker per task that needs doing; each worker can do its task and then exit.

Cheers --

Dave
  ----- Original Message ----- 
  From: Mike van Lammeren 
  To: FreeSWITCH Users 
  Sent: Friday, March 05, 2010 12:04 PM
  Subject: [Freeswitch-users] Which Lua script am I?


  Hello!


  I am working on a high-availability system and need a Lua script that can run in the background and loop forever, periodically waking up to accomplish some tasks. I would like to be able to handle at least 10 concurrent tasks.


  My first idea was to have a simple background script that would call another script, let's call it the 'task' script, every 5 seconds or so. This would spread the work around, and if any task script quit with an error, the background script would remain running, and would continue to launch new task scripts.


  However, I can see no way for the background script to launch other Lua scripts. Am I missing something? I need something like session:execute(), but unfortunately this background script has no session, and I'm not sure if session:execute() can even make a call like the "luarun" console command.


  So, my next ideas was to have 10 copies of the background script. Each script would loop forever. Upon launch, each script would perform a single task, then sleep for 60 seconds. This allows handling of 10 concurrent tasks.


  However, there are some problems with running 10 copies of the same script, which I believe can be solved if each script could be given a unique identifier. One problem is that I would like the scripts to actually sleep for a variable amount of time, say between 55 and 65 seconds, so that the workload would spread around. Another, similar problem, is that each script needs to 'claim' a number of database records from a table that is queuing up work for the scripts. Once again, if each script had a unique identifier, then I could use an update query to effectively 'set and test' a record that would be tied to that script instance.


  I could use a random number to identify each script, but since math.random is seeded with a timestamp that only resolves to seconds, then all 10 scripts will be given the same sequence of random numbers, and all 10 will run in lock-step with each other. So, random numbers are out. (If I had a unique ID to seed the random number generator, then I could just use that unique ID and wouldn't need random numbers.)


  Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH launches at startup?


  Or is there a better way of handling this situation altogether? I'm open to any suggestions that people would care to make!


  (I would entertain the idea of a single script that would start with FreeSWITCH, then launch 10 other scripts, giving each a unique identifier, but that brings me back to the original problem of not knowing how to launch other Lua scripts under FreeSWITCH.)


  Mike van Lammeren




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Fri Mar  5 11:42:54 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 13:42:54 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
Message-ID: <5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>

Write it in C and use the FreeSWITCH task scheduler API.

/b

On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:

> All the same, I'm still open to suggestions on better ways to achieve my ultimate goals!



From mike at van.lammeren.net  Fri Mar  5 11:59:48 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:59:48 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
Message-ID: <5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>

Thanks so much, everyone! Awesome advice, as usual!

Due to a deadline situation, I'm going with the Lua script idea. However, if
it gives us any trouble after the looming deadline, then I will definitely
go the C route.

Thanks again!

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:

> Write it in C and use the FreeSWITCH task scheduler API.
>
> /b
>
> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>
> > All the same, I'm still open to suggestions on better ways to achieve my
> ultimate goals!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at van.lammeren.net  Fri Mar  5 12:17:35 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 15:17:35 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
Message-ID: <5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>

In case anyone's interested, I just found something cool. You can pass
parameters to the Lua script from the lua.conf.xml file, like so:

    <param name="startup-script" value="temp.lua script1"/>
    <param name="startup-script" value="temp.lua script2"/>
    <param name="startup-script" value="temp.lua script3"/>

I can see that being useful.

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Thanks so much, everyone! Awesome advice, as usual!
>
> Due to a deadline situation, I'm going with the Lua script idea. However,
> if it gives us any trouble after the looming deadline, then I will
> definitely go the C route.
>
> Thanks again!
>
> Mike van Lammeren
>
>
> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Write it in C and use the FreeSWITCH task scheduler API.
>>
>> /b
>>
>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>>
>> > All the same, I'm still open to suggestions on better ways to achieve my
>> ultimate goals!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From rob4manhere at gmail.com  Fri Mar  5 12:31:35 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 14:31:35 -0600
Subject: [Freeswitch-users] ESL Event
Message-ID: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>

Hi all,

I'm sending a custom event via an outbound ESL socket (using ESL's  
pymod ESL.py and _ESL.so).

Does this look right?

         event = ESL.ESLevent("CUSTOM", "test::event1")
         event.addHeader("Name1", "Value1")
         self.esl.sendEvent(event)


It used to work (r16593) but now doesn't on trunk (r16897).  I have a  
lot more information which I'll submit in jira, but just wanted to  
check that I'm not missing something obvious about sending custom  
events before I do.

All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3 (stock).

Rob

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From brian at freeswitch.org  Fri Mar  5 12:39:56 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 14:39:56 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
Message-ID: <828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>

Wouldn't that be event.sendEvent?

/b

On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:

> 
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)

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From anthony.minessale at gmail.com  Fri Mar  5 12:43:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 14:43:26 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
Message-ID: <191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>

are you trying to send the event to a specific channel or fire the event?

The default action is fire the event, if you want to send it to a specific
channel you need to
add the unique-id header with the appropriate uuid.



On Fri, Mar 5, 2010 at 2:31 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> Hi all,
>
> I'm sending a custom event via an outbound ESL socket (using ESL's pymod
> ESL.py and _ESL.so).
>
> Does this look right?
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> It used to work (r16593) but now doesn't on trunk (r16897).  I have a lot
> more information which I'll submit in jira, but just wanted to check that
> I'm not missing something obvious about sending custom events before I do.
>
> All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3 (stock).
>
> Rob
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rob4manhere at gmail.com  Fri Mar  5 12:49:40 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 14:49:40 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>
Message-ID: <26BCAA97-D813-4C84-AEA2-3149EDB3CE84@gmail.com>

Sorry I didn't specify.

I'm trying to fire an event, which is picked up by another event  
listening daemon.


On Mar 5, 2010, at 2:43 PM, Anthony Minessale wrote:

> are you trying to send the event to a specific channel or fire the  
> event?
>
> The default action is fire the event, if you want to send it to a  
> specific channel you need to
> add the unique-id header with the appropriate uuid.
>
>
>
> On Fri, Mar 5, 2010 at 2:31 PM, Rob Forman <rob4manhere at gmail.com>  
> wrote:
> Hi all,
>
> I'm sending a custom event via an outbound ESL socket (using ESL's  
> pymod ESL.py and _ESL.so).
>
> Does this look right?
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> It used to work (r16593) but now doesn't on trunk (r16897).  I have  
> a lot more information which I'll submit in jira, but just wanted to  
> check that I'm not missing something obvious about sending custom  
> events before I do.
>
> All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3  
> (stock).
>
> Rob
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rob4manhere at gmail.com  Fri Mar  5 14:02:30 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 16:02:30 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
Message-ID: <926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>

I don't think so.  I see sendEvent under the ESLconnection class in  
ESL.py.  And it used to work.

Just doing a sanity check.  If it looks right I'll submit a jira ticket.



On Mar 5, 2010, at 2:39 PM, Brian West wrote:

> Wouldn't that be event.sendEvent?
>
> /b
>
> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>
>>
>>         event = ESL.ESLevent("CUSTOM", "test::event1")
>>         event.addHeader("Name1", "Value1")
>>         self.esl.sendEvent(event)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Fri Mar  5 14:57:23 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Mar 2010 14:57:23 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com><87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com><79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
	<191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
Message-ID: <15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>

Sorry, I just looked for diffs in the logs I had.  Thank you for all of your
good advice.  I do appreciate it!  I did try a snom 360 phone (but not a
polycomm) and it behaved the same, however, I did find that one of my
servers (with a 2-day newer FS version and different config) does not have
this problem.  So I will isolate the issue between my config or just upgrade
to the newer FS.
 
Thank You and Best Regards,
Jerry
 



  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, March 04, 2010 4:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialing **<extension>


Did you totally ignore my instructions?
Do you want help or not?



On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


What I see in the log where they first differ is just after FS sends the
CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
didn't work.  Do you know what PICKED_OFF means?
 
In the good case, FS logs:
 
tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities

In the bad case, FS logs:
 
tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
PICKED_OFF
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
<mailto:sofia/internal/1059 at 192.168.72.141:5060>
sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
<mailto:sofia/internal/5402 at 192.168.72.141:5060>
sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]

Best Regards,
Jerry


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Thursday, March 04, 2010 2:59 PM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Dialing **<extension>




On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry



I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC




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






-- 
Anthony Minessale II

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

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

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


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From anthony.minessale at gmail.com  Fri Mar  5 15:11:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 17:11:41 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
	<926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
Message-ID: <191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>

found your problem
fixed in r16921

I think in the future maybe just open the jira right away, it's easy to
close them as "not a bug" if you are wrong.
If you are right it spares us the trouble of tracking issues over email.


On Fri, Mar 5, 2010 at 4:02 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> I don't think so.  I see sendEvent under the ESLconnection class in ESL.py.
>  And it used to work.
>
> Just doing a sanity check.  If it looks right I'll submit a jira ticket.
>
>
>
> On Mar 5, 2010, at 2:39 PM, Brian West wrote:
>
> Wouldn't that be event.sendEvent?
>
> /b
>
> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar  5 15:15:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 17:15:06 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
	<79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
	<191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
	<15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>
Message-ID: <191c3a031003051515u232820a3ob8b13ab70377549c@mail.gmail.com>

What I am trying to tell you going forward is to open issues on jira not the
mailing list I have to say it 4-10 times a day to people.  We cant follow
issues over email, we get literally 100+ emails a day some days.  It's for
your own good.  We need proper workflow....



On Fri, Mar 5, 2010 at 4:57 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Sorry, I just looked for diffs in the logs I had.  Thank you for all of
> your good advice.  I do appreciate it!  I did try a snom 360 phone (but
> not a polycomm) and it behaved the same, however, I did find that one of my
> servers (with a 2-day newer FS version and different config) does not have
> this problem.  So I will isolate the issue between my config or just upgrade
> to the newer FS.
>
> Thank You and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Thursday, March 04, 2010 4:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>
> Did you totally ignore my instructions?
> Do you want help or not?
>
>
> On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>  What I see in the log where they first differ is just after FS sends the
>> CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
>> didn't work.  Do you know what PICKED_OFF means?
>>
>> In the good case, FS logs:
>>
>> tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060(already 1)
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
>> ORIGINATOR_CANCEL
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
>> sleep
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
>> entities
>> In the bad case, FS logs:
>>
>> tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060(already 1)
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
>> CS_REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
>> PICKED_OFF
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
>> sleep
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
>> entities
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
>> sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
>> [ORIGINATOR_CANCEL]
>> Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Michael Collins [mailto:msc at freeswitch.org]
>> *Sent:* Thursday, March 04, 2010 2:59 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>>
>>
>>
>> On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>
>>> By the way, it appears that the "**<extension>" ringing call pickup
>>> feature
>>> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
>>> of
>>> the following scenario:
>>>
>>> 1) 1059 calls 5381
>>> 2) 5381 rings
>>> 3) 5402 dials "**5381"
>>>
>>> In the bad case (http://pastebin.freeswitch.org/12325), all calls
>>> disconnect
>>> following above scenario.  In the good case
>>> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>>>
>>> Best Regards,
>>> Jerry
>>>
>>
>> I'd suggest turning off the sofia debug stuff cuz that's just line noise
>> unless you know all that stuff. You need to compare the two traces and look
>> for differences. I'd start by looking at only the fs console output of each
>> call and see if there's anything different between the two. See if there's a
>> point where they diverge and work your way back from there. Look at the
>> corresponding SIP dialogs and see if there are any clues as well. You may
>> also want to capture the SIP traffic with tcpdump and analyze it in
>> Wireshark which is easier on the eyes than having it buried in with the
>> sofia and console logs.
>>
>> FWIW, I just updated to latest and I can't make this feature *NOT* work.
>> I'm on 32-bit CentOS using Polycom 550 and Snom 320 phones.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rupa at rupa.com  Fri Mar  5 15:29:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Mar 2010 17:29:49 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
Message-ID: <b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>

hmm.. at some point postgres is falling behind and rather than waiting,
we're opening another connection.  <ponder/>  can you open a jira on this?
 we probably need to implement connection limits in the core db connection
thingie.

On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>        I found big odbc problem when i try to move to FS 1.0.5.
> in same server i'm testing 1.0.4 and 1.0.5 and same config
> 1. config  limit.conf.xml use odbc (i use postgresql)
>  <param name="odbc-dsn" value="tuxza:admin:xxx"/>
> 2. create dialplan in context public for test limit
>  <condition field="destination_number" expression="^(1111)$">
>       <action application="limit" data="$${domain} 0838833133 500"/>
>      <action application="answer"/>
>      <action application="sleep" data="60000"/>
> </condition>
> 3. use sipp generate test call
>   sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
> xx.xx.xx.xx:5080
>
> 1.0.4 work fine i found 2 connection in postgresql server
> but 1.0.5 use 1 connection for 1 calls. and get error when call over
> 100 calls. (postgresql maximum client connect is 100 )
>
> So i'm not sure its' BUG or not ?
>
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From lloyd.aloysius at gmail.com  Fri Mar  5 15:34:32 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 5 Mar 2010 18:34:32 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
Message-ID: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>

Hi All,

I follow all the steps in the wiki.

Multi-tenant work properly , I need to setup DNS SRV and SIP for each
Domain.

If I do not have the access the domains DNS how can I make Multi-tenant
work.

Thank you.
Lloyd
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From jerry.richards at teotech.com  Fri Mar  5 15:36:08 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Mar 2010 15:36:08 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com><835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org><ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
	<191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>
Message-ID: <BD3EC285AA0B4887B173E8AF99051693@greyhawk.tonecommander.com>

Sorry again for the late reply on this issue.  I think I understand what
you're saying.  Since IMs are not sent in the context of a dialog (it's
simply a send/receive handshake), the b2bua has nothing to tell the
originator.  I will investigate the possibility of getting the Bria phones
to send plain text.  Regarding commercial support options for FS, I raised
the issue to management here, but I'm not sure what the outcome of that will
be (it's not my decision to make).
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, February 26, 2010 12:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling


It says SHOULD, not MUST right?

The message passing in FS is abstracted and protocol agnostic and we are a
b2bua not a proxy in terms of SIP.
You are sending a message to 1 UA on FS who is accepting the message and
delivering it to the core who is happy to receive messages in any format.
Then it's routed back out another sip dialog where it's rejected.  It's too
late to go tell the sender that is unacceptable because we already happily
accepted it (messages are not always passed out to other phones they can
easily be directed at other internal resources).  We don't know what the
content-type means as we are a neutral party in the whole thing so there is
not much else we can do but violate this scope issue and break out of our
role as a neutral party and translate it to plain text and try again which
is not very elegant.

If FS was a proxy software, like openser and friends, we would be passing
the data between UA in the way you expect but we are not a proxy.


Based on the frequency and specific nature of your constant inquiries, and
the likelihood that you are offering commercial VoIP services to customers.
I suggest you contact us at consulting at freeswitch.org to investigate
commercial support options for FreeSWITCH.  Even then, I am not sure I could
help you besides maybe a param to convert all text/html messages to plain
text or some other sad hack.





On Fri, Feb 26, 2010 at 1:43 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I'm not sure I follow your comment.  The first device prefers text/html so
that's what it normally sets in the initial MESSAGE.  Devices that support
text/html will not generate this 415 error reply.  It's only devices that
don't support it that would send the 415 reply, so the issue is that the 415
is not getting passed back to the originator.

Best Regards,
Jerry



-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org]
Sent: Friday, February 26, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling

Its really clear here you'll need to say text/plain in the content type
their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

>
> I have two types of devices, one supports text/html MESSAGE content
> and one that only supports text/plain MESSAGE content.  When I send an
> IM from the first to the second, the second replies with 415
> Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261
> says the sender should retry using the media type acceptable to the
receiver (in this case: plain/text).
>
> The problem I have is that Freeswitch doesn't pass the error back to
> the sender (nor does it retry itself using plain/text).  So the IM is
lost.
> Does anyone see the reason why the error is not being handled correctly?
>
>
> ----------------------------------------------------------------------
> -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141
<mailto:sip%3A5382 at 192.168.72.141> >;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
>
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>
> ----------------------------------------------------------------------
> -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> .72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141
<mailto:sip%3A5382 at 192.168.72.141> >;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org





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





-- 
Anthony Minessale II

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

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

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


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From rob4manhere at gmail.com  Fri Mar  5 15:40:08 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 17:40:08 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
	<926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
	<191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>
Message-ID: <64C2AEA6-E0BA-483F-BAB8-1F85BBEC070A@gmail.com>

Thanks, and will do.


On Mar 5, 2010, at 5:11 PM, Anthony Minessale wrote:

> found your problem
> fixed in r16921
>
> I think in the future maybe just open the jira right away, it's easy  
> to close them as "not a bug" if you are wrong.
> If you are right it spares us the trouble of tracking issues over  
> email.
>
>
> On Fri, Mar 5, 2010 at 4:02 PM, Rob Forman <rob4manhere at gmail.com>  
> wrote:
> I don't think so.  I see sendEvent under the ESLconnection class in  
> ESL.py.  And it used to work.
>
> Just doing a sanity check.  If it looks right I'll submit a jira  
> ticket.
>
>
>
> On Mar 5, 2010, at 2:39 PM, Brian West wrote:
>
>> Wouldn't that be event.sendEvent?
>>
>> /b
>>
>> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>>
>>>
>>>         event = ESL.ESLevent("CUSTOM", "test::event1")
>>>         event.addHeader("Name1", "Value1")
>>>         self.esl.sendEvent(event)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Fri Mar  5 15:43:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Mar 2010 17:43:32 -0600
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
Message-ID: <b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>

You have access to your own DNS, so you could do something like:

companya.supersip.com
companyb.supersip.com
etc

so you are actually doing the dns in your domain (supersip.com) and
provision the phones as such.

On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I follow all the steps in the wiki.
>
> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
> Domain.
>
> If I do not have the access the domains DNS how can I make Multi-tenant
> work.
>
> Thank you.
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From anthony.minessale at gmail.com  Fri Mar  5 16:43:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 18:43:45 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
Message-ID: <191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>

Now you are getting a dedicated handle per thread because of our cache db
system.  Either allow a max connections that matches the most calls you
anticipate or file a bug with rupa to downgrade the mod back to sharing one
db handle globally.

On Mar 5, 2010 5:35 PM, "Rupa Schomaker" <rupa at rupa.com> wrote:

hmm.. at some point postgres is falling behind and rather than waiting,
we're opening another connection.  <ponder/>  can you open a jira on this?
 we probably need to implement connection limits in the core db connection
thingie.



On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>
> Dear All,
>        I...
-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From lloyd.aloysius at gmail.com  Fri Mar  5 17:32:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 5 Mar 2010 20:32:34 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
	<b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
Message-ID: <8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>

Thanks Rupa. I will try your suggestion.


On Fri, Mar 5, 2010 at 6:43 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> You have access to your own DNS, so you could do something like:
>
> companya.supersip.com
> companyb.supersip.com
> etc
>
> so you are actually doing the dns in your domain (supersip.com) and
> provision the phones as such.
>
> On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I follow all the steps in the wiki.
>>
>> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
>> Domain.
>>
>> If I do not have the access the domains DNS how can I make Multi-tenant
>> work.
>>
>> Thank you.
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Mar  5 19:28:41 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 6 Mar 2010 11:28:41 +0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
Message-ID: <23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>

and also you can get the script name from argv[0] if you name you
script in different names

2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
> In case anyone's interested, I just found something cool. You can pass
> parameters to the Lua script from the lua.conf.xml file, like so:
> ?? ?<param name="startup-script" value="temp.lua script1"/>
> ?? ?<param name="startup-script" value="temp.lua script2"/>
> ?? ?<param name="startup-script" value="temp.lua script3"/>
> I can see that being useful.
> Mike van Lammeren
>
> On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net>
> wrote:
>>
>> Thanks so much, everyone! Awesome advice, as usual!
>> Due to a deadline situation, I'm going with the Lua script idea. However,
>> if it gives us any trouble after the looming deadline, then I will
>> definitely go the C route.
>> Thanks again!
>> Mike van Lammeren
>>
>> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> Write it in C and use the FreeSWITCH task scheduler API.
>>>
>>> /b
>>>
>>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>>>
>>> > All the same, I'm still open to suggestions on better ways to achieve
>>> > my ultimate goals!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dome at tel.co.th  Fri Mar  5 19:36:07 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 10:36:07 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
	<191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>
Message-ID: <8ccbff061003051936ue24c8f9pda00418a36d93313@mail.gmail.com>

How to limit cache db max connections  in FS ?

Dome C.

2010/3/6 Anthony Minessale <anthony.minessale at gmail.com>:
> Now you are getting a dedicated handle per thread because of our cache db
> system.? Either allow a max connections that matches the most calls you
> anticipate or file a bug with rupa to downgrade the mod back to sharing one
> db handle globally.
>
> On Mar 5, 2010 5:35 PM, "Rupa Schomaker" <rupa at rupa.com> wrote:
>
> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another connection. ?<ponder/> ?can you open a jira on this?
> ?we probably need to implement connection limits in the core db connection
> thingie.
>
> On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ?I...
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dome at tel.co.th  Fri Mar  5 19:39:18 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 10:39:18 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
Message-ID: <8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>

Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
I'll open JIRA for this case.

Dome C.

2010/3/6 Rupa Schomaker <rupa at rupa.com>:
> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another connection. ?<ponder/> ?can you open a jira on this?
> ?we probably need to implement connection limits in the core db connection
> thingie.
>
> On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ?I found big odbc problem when i try to move to FS 1.0.5.
>> in same server i'm testing 1.0.4 and 1.0.5 and same config
>> 1. config ?limit.conf.xml use odbc (i use postgresql)
>> ?<param name="odbc-dsn" value="tuxza:admin:xxx"/>
>> 2. create dialplan in context public for test limit
>> ?<condition field="destination_number" expression="^(1111)$">
>> ? ? ? <action application="limit" data="$${domain} 0838833133 500"/>
>> ? ? ?<action application="answer"/>
>> ? ? ?<action application="sleep" data="60000"/>
>> </condition>
>> 3. use sipp generate test call
>> ? sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
>> xx.xx.xx.xx:5080
>>
>> 1.0.4 work fine i found 2 connection in postgresql server
>> but 1.0.5 use 1 connection for 1 calls. and get error when call over
>> 100 calls. (postgresql maximum client connect is 100 )
>>
>> So i'm not sure its' BUG or not ?
>>
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Fri Mar  5 22:24:00 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 01:24:00 -0500
Subject: [Freeswitch-users] wiki DNS page update
Message-ID: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>

for anyone who can update this page for NAPTR ex

http://wiki.freeswitch.org/wiki/DNS


# NAPTR for SIP over UDP
example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
# NAPTR for SIP over TCP
example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
# NAPTR for SIP over TLS
example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

Regards

Franck


-- 
This message has been scanned for viruses and
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From infos at madovsky.org  Fri Mar  5 23:28:45 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 02:28:45 -0500
Subject: [Freeswitch-users] fundamental question
Message-ID: <DD4280539BDB4AEB9FCC25ECA7BD2CF8@MOBILEE1705>

Concerning softphones not SRV compatible
how to make them work for registration with a domain name managed by NAPTR and/or SRV ?
I tried 

x-lite  --->  register
3cx    ---> not register

Thanks

Franck

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From brian at freeswitch.org  Sat Mar  6 07:42:50 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 09:42:50 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
Message-ID: <709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>

Anyone can update it thats the point.  Login and add them.

http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS

Verify them against that.  I think the flag is lowercase.

/b

On Mar 6, 2010, at 12:24 AM, Madovsky wrote:

> for anyone who can update this page for NAPTR ex
>  
> http://wiki.freeswitch.org/wiki/DNS
>  
>  
> # NAPTR for SIP over UDP
> example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
> # NAPTR for SIP over TCP
> example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
> # NAPTR for SIP over TLS
> example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.
>  
> Regards
>  
> Franck
> 
> 
> -- 
> This message has been scanned for viruses and 
> dangerous content by MailScanner, and is 
> believed to be clean. _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar  6 08:24:28 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 11:24:28 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
Message-ID: <64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>

It doesn't matter apparently,
but it needs to correct the line

# NAPTR for SIP over TLS
example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 10:42 AM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Anyone can update it thats the point.  Login and add them.


  http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


  Verify them against that.  I think the flag is lowercase.


  /b


  On Mar 6, 2010, at 12:24 AM, Madovsky wrote:


    for anyone who can update this page for NAPTR ex

    http://wiki.freeswitch.org/wiki/DNS


    # NAPTR for SIP over UDP
    example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
    # NAPTR for SIP over TCP
    example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

    Regards

    Franck



    -- 
    This message has been scanned for viruses and 
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    believed to be clean. _______________________________________________
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From infos at madovsky.org  Sat Mar  6 08:25:26 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 11:25:26 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
Message-ID: <551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>

but anyway, to be sure let's change it in lower case
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 10:42 AM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Anyone can update it thats the point.  Login and add them.


  http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


  Verify them against that.  I think the flag is lowercase.


  /b


  On Mar 6, 2010, at 12:24 AM, Madovsky wrote:


    for anyone who can update this page for NAPTR ex

    http://wiki.freeswitch.org/wiki/DNS


    # NAPTR for SIP over UDP
    example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
    # NAPTR for SIP over TCP
    example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

    Regards

    Franck



    -- 
    This message has been scanned for viruses and 
    dangerous content by MailScanner, and is 
    believed to be clean. _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sat Mar  6 09:02:44 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:02:44 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
	<64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
Message-ID: <427E641C-70AA-457A-80EA-675360805515@freeswitch.org>

I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.

/b

On Mar 6, 2010, at 10:24 AM, Madovsky wrote:

> # NAPTR for SIP over TLS
> example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.
> 

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From brian at freeswitch.org  Sat Mar  6 09:09:09 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:09:09 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
	<551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
Message-ID: <6F12E9C2-2C42-421E-89D8-139FD630D1BB@freeswitch.org>

I don't know who changed it to  _sips._tcp.domain.com. but that is 100% WRONG I want back to my zone files I used for testing and its _sip._tls and you can even google that.  

Can someone pin down something to prove me wrong?

/b

On Mar 6, 2010, at 10:25 AM, Madovsky wrote:

> but anyway, to be sure let's change it in lower case
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 10:42 AM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> Anyone can update it thats the point.  Login and add them.
> 
> http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS
> 
> Verify them against that.  I think the flag is lowercase.
> 
> /b

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From infos at madovsky.org  Sat Mar  6 09:11:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:11:05 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
Message-ID: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>

I dont' think _sip._tls is standard, but if it works, great
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:02 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.


  /b


  On Mar 6, 2010, at 10:24 AM, Madovsky wrote:


    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.






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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sat Mar  6 09:15:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:15:05 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
	<8D677CB395454F21B5691A27515AB558@MOBILEE1705>
Message-ID: <776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>

I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls

So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.

/b

On Mar 6, 2010, at 11:11 AM, Madovsky wrote:

> I dont' think _sip._tls is standard, but if it works, great
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:02 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.
> 
> /b

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From brian at freeswitch.org  Sat Mar  6 09:15:24 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:15:24 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
	<8D677CB395454F21B5691A27515AB558@MOBILEE1705>
Message-ID: <30FD929F-6817-450D-A240-E4AB742D04D9@freeswitch.org>

Oh and eyebeam does _sip._tls

/b

On Mar 6, 2010, at 11:11 AM, Madovsky wrote:

> I dont' think _sip._tls is standard, but if it works, great
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:02 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.
> 
> /b

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From infos at madovsky.org  Sat Mar  6 09:35:54 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:35:54 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
	<6F12E9C2-2C42-421E-89D8-139FD630D1BB@freeswitch.org>
Message-ID: <21F56D3A23CF434AA7E2844256AD0494@MOBILEE1705>

What DNS server you use ?
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:09 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I don't know who changed it to  _sips._tcp.domain.com. but that is 100% WRONG I want back to my zone files I used for testing and its _sip._tls and you can even google that.  


  Can someone pin down something to prove me wrong?


  /b


  On Mar 6, 2010, at 10:25 AM, Madovsky wrote:


    but anyway, to be sure let's change it in lower case
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 10:42 AM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      Anyone can update it thats the point.  Login and add them.


      http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


      Verify them against that.  I think the flag is lowercase.


      /b




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From infos at madovsky.org  Sat Mar  6 09:36:50 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:36:50 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705>
	<776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>
Message-ID: <B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>

ok
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:15 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls


  So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.


  /b


  On Mar 6, 2010, at 11:11 AM, Madovsky wrote:


    I dont' think _sip._tls is standard, but if it works, great
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 12:02 PM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.


      /b




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From brian at freeswitch.org  Sat Mar  6 10:05:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 12:05:48 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705>
	<776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>
	<B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
Message-ID: <46F21E0D-FDEC-49C4-9725-2B14C12CDF42@freeswitch.org>

Let clarify this ... after digging thru the code sofia looks for _sips._tcp but polycom/snom/eyebeam will look for _sip._tls, 

I use Bind.

/b

On Mar 6, 2010, at 11:36 AM, Madovsky wrote:

> ok
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:15 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls
> 
> So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.
> 
> /b

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From anthony.minessale at gmail.com  Sat Mar  6 10:06:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Mar 2010 12:06:29 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>
	<191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
Message-ID: <191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>

Did anyone read my last email?

It is only worthy of jira if its a real problem.
There is 1 db con per channel did you try increasing the max conns and
compare the performance?

On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
I'll open JIRA for this case.

Dome C.

2010/3/6 Rupa Schomaker <rupa at rupa.com>:

> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another co...
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From infos at madovsky.org  Sat Mar  6 10:17:14 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 13:17:14 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705><776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org><B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
	<46F21E0D-FDEC-49C4-9725-2B14C12CDF42@freeswitch.org>
Message-ID: <0B32A78695AA4AF19FF2E286B5F29D2B@MOBILEE1705>

eyebeam doesn't use strct standard stuff
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 1:05 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Let clarify this ... after digging thru the code sofia looks for _sips._tcp but polycom/snom/eyebeam will look for _sip._tls, 


  I use Bind.


  /b


  On Mar 6, 2010, at 11:36 AM, Madovsky wrote:


    ok
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 12:15 PM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls


      So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.


      /b




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From infos at madovsky.org  Sat Mar  6 10:19:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 13:19:10 -0500
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk
	16912)Problem.
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com><b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com><8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com><191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
	<191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
Message-ID: <48640F1B60F64D87BCF88646668EB341@MOBILEE1705>

I use postgresql, but I can test it if you have a tools that
similate calls charge.
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 1:06 PM
  Subject: Re: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)Problem.


  Did anyone read my last email?

  It is only worthy of jira if its a real problem. 
  There is 1 db con per channel did you try increasing the max conns and compare the performance?



    On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

    Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
    I'll open JIRA for this case.

    Dome C.

    2010/3/6 Rupa Schomaker <rupa at rupa.com>:

    > hmm.. at some point postgres is falling behind and rather than waiting,
    > we're opening another co...




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From dome at tel.co.th  Sat Mar  6 19:54:48 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 7 Mar 2010 10:54:48 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>
	<191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
	<191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
Message-ID: <8ccbff061003061954y16da64e1m16e17a743b943490@mail.gmail.com>

2010/3/7 Anthony Minessale <anthony.minessale at gmail.com>:
> Did anyone read my last email?

Sorry just read it:)

>
> It is only worthy of jira if its a real problem.
> There is 1 db con per channel did you try increasing the max conns and
> compare the performance?
max conns ? you mean in postgresql ? I think postgresql need more
resource if change max client  from 100 to 2000

Now i port mod_limit from 1.0.4 to 1.0.5 it's work for this case. but
i'm not sure about other db connect (core , sofia)
i only use mod_limit.

>
> On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:
>
> Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
> I'll open JIRA for this case.
>
> Dome C.
>
> 2010/3/6 Rupa Schomaker <rupa at rupa.com>:
>
>> hmm.. at some point postgres is falling behind and rather than waiting,
>> we're opening another co...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From shaheryarkh at googlemail.com  Sun Mar  7 01:38:44 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 7 Mar 2010 14:38:44 +0500
Subject: [Freeswitch-users]
	=?utf-8?q?Tienes_una_invitaci=C3=B3n_a_ClubWNC?=
In-Reply-To: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>
References: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>
Message-ID: <b16156851003070138x6ba2e08dred8c2d6daeb5e2e2@mail.gmail.com>

Hola,

Chicos, por favor, poner fin a este correo spam en list.I FreeSwitch estoy
haciendo unos 20-30 mensajes de este tipo todos los d?as. S?rvase no abusen
de este foro para fines publicitarios y de marketing.

Gracias.


2010/3/4 seven <dujinfang at gmail.com>

> [image: ClubWNC]<http://www.clubwnc.com/eshop/index.php?sponsor_id=0&cdr=2>
>  *Tienes una invitaci?n a ClubWNC*
>  Hola !!!
> He creado un perfil en ClubWNC donde puedo hacer listas de regalos,
> publicar rese?as sobre productos, ver a mis amigos de Facebook y lo que les
> gusta, lo que han comprado, y lo que quieren que les regalen, y quiero
> agregarte a mis amigos para que puedas verlo. Para ello, necesitas
> registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.
> Para registrarte en ClubWNC, sigue este enlace:
> http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895<http://www.clubwnc.com/eshop/create_eshop_account.php?sponsor_id=0&cdr=2895>     seven
> ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si
> no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para
> borrar tu nombre de la lista de personas suscritas.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From neil.burgess at redmatter.com  Sun Mar  7 04:31:31 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Sun, 7 Mar 2010 12:31:31 +0000
Subject: [Freeswitch-users] Export b-leg variables in bridge string for a
	user bridge lookup
Message-ID: <787302A89ACCE24DA8F56DA101E77C84139B8D275D@THHS2E12BE1X.hostedservice2.net>

Hi,

Wonder if anyone can help me with a bridge to user question?

We are trying to bridge to a user in the directory using a command of the form:-

session.execute("bridge","[leg_confirm=yes]user/1009");

We want to be able to pass a number of channel variables to the b-leg. In the case above, we want to pass leg_confirm to enable us to use the group_confirm functionality on the leg, but have found that whilst this is fine using something like session.execute("bridge","[leg_confirm=yes]sofia/blah/blah"), when used against a user/  lookup it doesn't set the variable.

I can see that the dial-string in the User's Directory definition would presumably need these bridge channel variables added to it somehow, but don't see how to achieve that!

We have used export in some cases to set the channel variable on the b-leg for a user which is fine, until we want to do something like session.execute("bridge","[leg_confirm=yes]user/1009, [leg_confirm=no]user/1012"), where the two b-legs have differing requirements.

Any ideas how I can achieve a working version of session.execute("bridge","[leg_confirm=yes]user/1009");

As an added twist, we are using mod_xml_curl to perform our directory lookups!

Many thanks,
Neil Burgess

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From infos at madovsky.org  Sun Mar  7 10:56:42 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 7 Mar 2010 13:56:42 -0500
Subject: [Freeswitch-users] xmpp IM
Message-ID: <660B041BADA44ADE924CD4E293B118BB@MOBILEE1705>

Is there any already existing vas that contains IM in mod_dingaling ?

Thanks

Franck
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From paul.gore.j at gmail.com  Sun Mar  7 11:03:46 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 14:03:46 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
Message-ID: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>

Greetings,
We use XML curl to fetch our dial plan extensions from a database. As per FS
wiki we can only return one XML extension upon FS request.
Now we need to return a set of extensions with same number condition and
context but with different second condition so that FS can pick up the right
one.
Is it possible at all? And if yes how response XML with multiple extensions
should look like?

Thanks a lot.
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From lloyd.aloysius at gmail.com  Sun Mar  7 11:55:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 14:55:50 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
	<b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
	<8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>
Message-ID: <8a19bf2e1003071155m76bbc0kee7688929604cbcc@mail.gmail.com>

Rupa,

Thank you again. It is working great.

Lloyd


On Fri, Mar 5, 2010 at 8:32 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thanks Rupa. I will try your suggestion.
>
>
>
> On Fri, Mar 5, 2010 at 6:43 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> You have access to your own DNS, so you could do something like:
>>
>> companya.supersip.com
>> companyb.supersip.com
>> etc
>>
>> so you are actually doing the dns in your domain (supersip.com) and
>> provision the phones as such.
>>
>> On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>>
>>> Hi All,
>>>
>>> I follow all the steps in the wiki.
>>>
>>> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
>>> Domain.
>>>
>>> If I do not have the access the domains DNS how can I make Multi-tenant
>>> work.
>>>
>>> Thank you.
>>> Lloyd
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Sun Mar  7 12:03:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 15:03:34 -0500
Subject: [Freeswitch-users] Voice Mail 2 Email Configuration
Message-ID: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>

Hi All,

I am running FreeSWITCH on CentOS5.4 . Sendmail already installed in the
system and running . I can send email the following way

echo "test" | mail -s testsubject lloyd.aloysius at gmail.com

Here is the FreeSWITCH configuration

*1. Extension setup*

<include>
  <user id="201">
    <params>
      <param name="password" value="201"/>
      <param name="vm-password" value="201"/>
        <param name="vm-email-all-messages" value="true"/>
      <param name="vm-attach-file" value="true" />
      <param name="vm-mailto" value="lloyd.aloysius at gmail.com"/>

    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="201"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 201"/>
      <variable name="effective_caller_id_number" value="201"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


*2. /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml*


   <param name="mailer-app" value="sendmail"/>
    <param name="mailer-app-args" value="-t"/>
    <param name="dump-cores" value="yes"/>

But not a single voicemail deliver to email. How to troubleshoot this
problem. Am I missing any configuration?


Thanks,
Lloyd
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From nazim.agabekov at gmail.com  Sun Mar  7 12:18:07 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Mon, 08 Mar 2010 00:18:07 +0400
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
Message-ID: <4B9409FF.5060608@gmail.com>

Hello Paul

I also use mod_xml_curl for a dialplan. I've solved this issue by
generating the XML extension dynamically.
My fastcgi daemon checks the input variables like caller id, originator,
e.t.c, makes routing decision
and dynamically generates only one XML extension.
So, there is no  real need for creating complex dialplan on mod_xml_curl
output.

Nazim

On 03/07/2010 11:03 PM, paul gore wrote:
> Greetings,
> We use XML curl to fetch our dial plan extensions from a database. As
> per FS wiki we can only return one XML extension upon FS request. 
> Now we need to return a set of extensions with same number condition
> and context but with different second condition so that FS can pick up
> the right one. 
> Is it possible at all? And if yes how response XML with multiple
> extensions should look like?
>
> Thanks a lot.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From lloyd.aloysius at gmail.com  Sun Mar  7 12:36:07 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 15:36:07 -0500
Subject: [Freeswitch-users] Voice Mail 2 Email Configuration
In-Reply-To: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>
References: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>
Message-ID: <8a19bf2e1003071236n67f7f2d9l9c7a234ae1bb4fe2@mail.gmail.com>

Hi All,

I am answering my question. I fixed the problem it is a sendemail problem.

Thanks
Lloyd


On Sun, Mar 7, 2010 at 3:03 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I am running FreeSWITCH on CentOS5.4 . Sendmail already installed in the
> system and running . I can send email the following way
>
> echo "test" | mail -s testsubject lloyd.aloysius at gmail.com
>
> Here is the FreeSWITCH configuration
>
> *1. Extension setup*
>
> <include>
>   <user id="201">
>     <params>
>       <param name="password" value="201"/>
>       <param name="vm-password" value="201"/>
>         <param name="vm-email-all-messages" value="true"/>
>       <param name="vm-attach-file" value="true" />
>       <param name="vm-mailto" value="lloyd.aloysius at gmail.com"/>
>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="201"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 201"/>
>       <variable name="effective_caller_id_number" value="201"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
>
> *2. /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml*
>
>
>    <param name="mailer-app" value="sendmail"/>
>     <param name="mailer-app-args" value="-t"/>
>     <param name="dump-cores" value="yes"/>
>
> But not a single voicemail deliver to email. How to troubleshoot this
> problem. Am I missing any configuration?
>
>
> Thanks,
> Lloyd
>
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From m.sobkow at marketelsystems.com  Sun Mar  7 13:33:03 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sun, 07 Mar 2010 15:33:03 -0600
Subject: [Freeswitch-users] Having some trouble recording calls.
Message-ID: <4B941B8F.1070506@marketelsystems.com>

I'm using the uuid_record command to tell Freeswitch to record a call.

    RecordResult = pbx:api( uuid_record, 
Operator#pbx_operator_registry.operator_uuid ++ " start " ++ FullFileName ),

After the customer hangs up, I tell it to stop recording:

    StopRecordResult = pbx:api( uuid_record, 
Operator#pbx_operator_registry.operator_uuid ++ " stop " ++ FullFileName ),

However, I get an error "-ERR Cannot stop record session!".  Any 
subsequent attempts to start a new recording file also fail.

What am I doing wrong?


From m.sobkow at marketelsystems.com  Sun Mar  7 14:10:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sun, 07 Mar 2010 16:10:10 -0600
Subject: [Freeswitch-users] Having some trouble recording calls.
In-Reply-To: <4B941B8F.1070506@marketelsystems.com>
References: <4B941B8F.1070506@marketelsystems.com>
Message-ID: <4B942442.2050001@marketelsystems.com>

Mark Sobkow wrote:
> I'm using the uuid_record command to tell Freeswitch to record a call.
>
>    RecordResult = pbx:api( uuid_record, 
> Operator#pbx_operator_registry.operator_uuid ++ " start " ++ 
> FullFileName ),
>
> After the customer hangs up, I tell it to stop recording:
>
>    StopRecordResult = pbx:api( uuid_record, 
> Operator#pbx_operator_registry.operator_uuid ++ " stop " ++ 
> FullFileName ),
>
> However, I get an error "-ERR Cannot stop record session!".  Any 
> subsequent attempts to start a new recording file also fail.
>
> What am I doing wrong?
>
Actually the logs show a bit more.

The first time I try to record a session, I get an ok message.  When I 
stop that recording, it tells me ok as well.

But on every subsequent attempt to record part of the session, I'm 
getting errors.  It's as if Freeswitch will only allow you to record 
once for an operator session.


From paul.gore.j at gmail.com  Sun Mar  7 14:13:20 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 17:13:20 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <4B9409FF.5060608@gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
Message-ID: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>

Thanks fro the input.
I thought about such approach, the only problem I see is there that
I essentially would have to duplicate FS dial plan parsing functionality, so
my question basically is there a way to avoid that?
I have seen a Ruby dial plan example on XML curl wiki page which actually
seem to generate response with multiple extensions, but unfortunately FS
cannot read such response.

On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  Hello Paul
>
> I also use mod_xml_curl for a dialplan. I've solved this issue by
> generating the XML extension dynamically.
> My fastcgi daemon checks the input variables like caller id, originator,
> e.t.c, makes routing decision
> and dynamically generates only one XML extension.
> So, there is no  real need for creating complex dialplan on mod_xml_curl
> output.
>
> Nazim
>
>
> On 03/07/2010 11:03 PM, paul gore wrote:
>
> Greetings,
> We use XML curl to fetch our dial plan extensions from a database. As per
> FS wiki we can only return one XML extension upon FS request.
> Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?
>
>  Thanks a lot.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sun Mar  7 14:38:05 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 7 Mar 2010 14:38:05 -0800
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
Message-ID: <81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>

Can you pastebin a sample dialplan and script? I'd like to better  
visualize what problem you're trying to solve.
-MC

Sent from my iPhone

On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:

> Thanks fro the input.
> I thought about such approach, the only problem I see is there that  
> I essentially would have to duplicate FS dial plan parsing  
> functionality, so my question basically is there a way to avoid that?
> I have seen a Ruby dial plan example on XML curl wiki page which  
> actually seem to generate response with multiple extensions, but  
> unfortunately FS cannot read such response.
>
> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <nazim.agabekov at gmail.com 
> > wrote:
> Hello Paul
>
> I also use mod_xml_curl for a dialplan. I've solved this issue by  
> generating the XML extension dynamically.
> My fastcgi daemon checks the input variables like caller id,  
> originator, e.t.c, makes routing decision
> and dynamically generates only one XML extension.
> So, there is no  real need for creating complex dialplan on  
> mod_xml_curl output.
>
> Nazim
>
>
> On 03/07/2010 11:03 PM, paul gore wrote:
>> Greetings,
>> We use XML curl to fetch our dial plan extensions from a database.  
>> As per FS wiki we can only return one XML extension upon FS request.
>> Now we need to return a set of extensions with same number  
>> condition and context but with different second condition so that  
>> FS can pick up the right one.
>> Is it possible at all? And if yes how response XML with multiple  
>> extensions should look like?
>>
>> Thanks a lot.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From paul.gore.j at gmail.com  Sun Mar  7 15:07:27 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 18:07:27 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
Message-ID: <da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>

Well, basically we used to return XML dial plan response with one extension
like

<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="test9">
        <condition field="destination_number" expression="^83789$">
          <action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/>
        </condition>
      </extension></context></section></document>

and now we want to return response like that

<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="test9"><condition field="username"
expression="^756755$"/><condition field="destination_number"
expression="^83789$"><action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/></condition></extension>

<extension name="test9"><condition field="username"
expression="^5564645$"/><condition field="destination_number"
expression="^83789$"><action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/></condition></extension></context></section></document>


But FS seem not to understand the latter, we get "No route..."


On Sun, Mar 7, 2010 at 5:38 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Can you pastebin a sample dialplan and script? I'd like to better visualize
> what problem you're trying to solve.
> -MC
>
> Sent from my iPhone
>
> On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:
>
> Thanks fro the input.
> I thought about such approach, the only problem I see is there that
> I essentially would have to duplicate FS dial plan parsing functionality, so
> my question basically is there a way to avoid that?
> I have seen a Ruby dial plan example on XML curl wiki page which actually
> seem to generate response with multiple extensions, but unfortunately FS
> cannot read such response.
>
> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <<nazim.agabekov at gmail.com>
> nazim.agabekov at gmail.com> wrote:
>
>>  Hello Paul
>>
>> I also use mod_xml_curl for a dialplan. I've solved this issue by
>> generating the XML extension dynamically.
>> My fastcgi daemon checks the input variables like caller id, originator,
>> e.t.c, makes routing decision
>> and dynamically generates only one XML extension.
>> So, there is no  real need for creating complex dialplan on mod_xml_curl
>> output.
>>
>> Nazim
>>
>>
>> On 03/07/2010 11:03 PM, paul gore wrote:
>>
>> Greetings,
>> We use XML curl to fetch our dial plan extensions from a database. As per
>> FS wiki we can only return one XML extension upon FS request.
>> Now we need to return a set of extensions with same number condition and
>> context but with different second condition so that FS can pick up the right
>> one.
>> Is it possible at all? And if yes how response XML with multiple
>> extensions should look like?
>>
>>  Thanks a lot.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From paul.gore.j at gmail.com  Sun Mar  7 16:19:31 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 19:19:31 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
	<da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>
Message-ID: <da4ab26e1003071619w4851481ct1c322e98ae099e0d@mail.gmail.com>

Actually to be correct - FS does read such response, but looks like it
processes only the first one from the top, and if its condition evaluates to
false it does not process further extensions.

On Sun, Mar 7, 2010 at 6:07 PM, paul gore <paul.gore.j at gmail.com> wrote:

> Well, basically we used to return XML dial plan response with one extension
> like
>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>     <context name="default">
>       <extension name="test9">
>         <condition field="destination_number" expression="^83789$">
>           <action application="bridge" data="iax/guest at conference.freeswitch.org/888"/>
>         </condition>
>       </extension></context></section></document>
>
> and now we want to return response like that
>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>     <context name="default">
>       <extension name="test9"><condition field="username" expression="^756755$"/><condition field="destination_number" expression="^83789$"><action application="bridge" data="iax/guest at conference.freeswitch.org/888"/></condition></extension>
>
> <extension name="test9"><condition field="username" expression="^5564645$"/><condition field="destination_number" expression="^83789$"><action application="bridge" data="iax/guest at conference.freeswitch.org/888"/></condition></extension></context></section></document>
>
>
> But FS seem not to understand the latter, we get "No route..."
>
>
> On Sun, Mar 7, 2010 at 5:38 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Can you pastebin a sample dialplan and script? I'd like to better
>> visualize what problem you're trying to solve.
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:
>>
>> Thanks fro the input.
>> I thought about such approach, the only problem I see is there that
>> I essentially would have to duplicate FS dial plan parsing functionality, so
>> my question basically is there a way to avoid that?
>> I have seen a Ruby dial plan example on XML curl wiki page which actually
>> seem to generate response with multiple extensions, but unfortunately FS
>> cannot read such response.
>>
>> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <<nazim.agabekov at gmail.com>
>> nazim.agabekov at gmail.com> wrote:
>>
>>>  Hello Paul
>>>
>>> I also use mod_xml_curl for a dialplan. I've solved this issue by
>>> generating the XML extension dynamically.
>>> My fastcgi daemon checks the input variables like caller id, originator,
>>> e.t.c, makes routing decision
>>> and dynamically generates only one XML extension.
>>> So, there is no  real need for creating complex dialplan on mod_xml_curl
>>> output.
>>>
>>> Nazim
>>>
>>>
>>> On 03/07/2010 11:03 PM, paul gore wrote:
>>>
>>> Greetings,
>>> We use XML curl to fetch our dial plan extensions from a database. As per
>>> FS wiki we can only return one XML extension upon FS request.
>>> Now we need to return a set of extensions with same number condition and
>>> context but with different second condition so that FS can pick up the right
>>> one.
>>> Is it possible at all? And if yes how response XML with multiple
>>> extensions should look like?
>>>
>>>  Thanks a lot.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From josh at radianttiger.com  Sun Mar  7 16:28:31 2010
From: josh at radianttiger.com (Josh Rivers)
Date: Sun, 7 Mar 2010 16:28:31 -0800
Subject: [Freeswitch-users] Conference Recording
Message-ID: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>

On last week's conference call, someone was recording, did that recording
get put online somewhere?

Josh
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From brian at freeswitch.org  Sun Mar  7 16:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 18:32:28 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
Message-ID: <5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>

You generate exactly ONE extension which was the exact way it was designed to be used.  You don't have to parse anything only look at the variables you care about and run with it.  Its nothing too complex or overly intensive to do.  

/b

On Mar 7, 2010, at 4:13 PM, paul gore wrote:

> Thanks fro the input.
> I thought about such approach, the only problem I see is there that I essentially would have to duplicate FS dial plan parsing functionality, so my question basically is there a way to avoid that? 
> I have seen a Ruby dial plan example on XML curl wiki page which actually seem to generate response with multiple extensions, but unfortunately FS cannot read such response.



From paul.gore.j at gmail.com  Sun Mar  7 17:30:02 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 20:30:02 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
Message-ID: <da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>

OK, so that means multiple extensions not possible.
Yes, it's not overly intensive, but nevertheless I actually have to parse
regex in the second condition to find out which extension matches incoming
number and I wanted instead to let FS do that for me.
It is a bit complex since we store complete extension XML in DB, so I have
to retrieve all the extensions matching first condition, extract their
second condition, then run them in loop... Looks like I can be better off
using JS for that instead.
Thank you everybody.

On Sun, Mar 7, 2010 at 7:32 PM, Brian West <brian at freeswitch.org> wrote:

> You generate exactly ONE extension which was the exact way it was designed
> to be used.  You don't have to parse anything only look at the variables you
> care about and run with it.  Its nothing too complex or overly intensive to
> do.
>
> /b
>
> On Mar 7, 2010, at 4:13 PM, paul gore wrote:
>
> > Thanks fro the input.
> > I thought about such approach, the only problem I see is there that I
> essentially would have to duplicate FS dial plan parsing functionality, so
> my question basically is there a way to avoid that?
> > I have seen a Ruby dial plan example on XML curl wiki page which actually
> seem to generate response with multiple extensions, but unfortunately FS
> cannot read such response.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at gmail.com  Sun Mar  7 17:40:45 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Sun, 7 Mar 2010 17:40:45 -0800
Subject: [Freeswitch-users] mod xml rpc
Message-ID: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>

I just enabled mod xml rpc to test the web interface to retrieve the
voicemail; however, every time when I tried to access via
http://(fs_ip):8080/api/voicemail/web<http://%28fs_ip%29:8080/api/voicemail/web>,
somehow FS process got killed by itself.
I wonder whether anyone experienced this problem. I am running r16928.

Thank you,
Dorn B.
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From brian at freeswitch.org  Sun Mar  7 17:46:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 19:46:33 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
Message-ID: <84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>

Apparently you haven't used a SQL db before.  SELECT * from extensions where did='1234';  You can query for EXACTLY the info you need.

/b

On Mar 7, 2010, at 7:30 PM, paul gore wrote:

> It is a bit complex since we store complete extension XML in DB, so I have to retrieve all the extensions matching first condition, extract their second condition, then run them in loop... Looks like I can be better off using JS for that instead.



From brian at freeswitch.org  Sun Mar  7 17:47:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 19:47:12 -0600
Subject: [Freeswitch-users] mod xml rpc
In-Reply-To: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>
References: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>
Message-ID: <63AEAD12-F18F-4E61-B753-5F285ED067BB@freeswitch.org>

Please read this http://wiki.freeswitch.org/wiki/Reporting_Bugs and report a bug to jira.

/b

On Mar 7, 2010, at 7:40 PM, DJB INTERNATIONAL wrote:

> I just enabled mod xml rpc to test the web interface to retrieve the voicemail; however, every time when I tried to access via http://(fs_ip):8080/api/voicemail/web, somehow FS process got killed by itself.  
> I wonder whether anyone experienced this problem.  I am running r16928.
> 
> 
> Thank you,
> Dorn B.

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From paul.gore.j at gmail.com  Sun Mar  7 18:03:14 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 21:03:14 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
Message-ID: <da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>

:-))
 I used it a lot. Unfortunately we do not store did number as a field, as I
said we store complete extension XML, with extension number and context as a
key.
So essentially I will retrieve a bunch of extension XMLs which I need to
parse and run against regex matcher to find out the one, which is what
exactly FS is doing when XML dial plan is defined in a static file.



On Sun, Mar 7, 2010 at 8:46 PM, Brian West <brian at freeswitch.org> wrote:

> Apparently you haven't used a SQL db before.  SELECT * from extensions
> where did='1234';  You can query for EXACTLY the info you need.
>
> /b
>
> On Mar 7, 2010, at 7:30 PM, paul gore wrote:
>
> > It is a bit complex since we store complete extension XML in DB, so I
> have to retrieve all the extensions matching first condition, extract their
> second condition, then run them in loop... Looks like I can be better off
> using JS for that instead.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Mar  7 18:07:59 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 7 Mar 2010 21:07:59 -0500
Subject: [Freeswitch-users] alias in profile
Message-ID: <16B4AB8FDE0149A98BFA5128C312957B@MOBILEE1705>

Is anyone can enlight me of the definition
of "alias" in profile and for what is it made for ?

Thanks

Franck
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From brian at freeswitch.org  Sun Mar  7 18:17:19 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 20:17:19 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
Message-ID: <A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>

Sounds like you zigged where you should have zagged.

/b

On Mar 7, 2010, at 8:03 PM, paul gore wrote:

> :-))
>  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.



From rob4manhere at gmail.com  Sun Mar  7 18:34:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Sun, 7 Mar 2010 20:34:00 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
Message-ID: <e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>

Nice to see so many people using it in so many different ways.  For those
using ec2 or xen, have you noticed this in your logs?

2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating timer,
please wait...
2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer resolution of
4000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may experience
audio problems.


"You may experience audio problems."  Seems pretty clear.  Can that just be
ignored?

Rob


On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
> wrote:

> I used to do tests on conferencing using linode.com, which is Xen VMs
> without issues, although there were not much load on it.
>
> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
>> pretty well. Of course no much load.
>>
>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
>> wrote:
>> >>  >  Cheers, Chris.
>> >>  >  --
>> >>  >  RightScale, Inc.
>> >>
>> >> I doubt they'll be switching any time soon.  I think his employer IS
>> >> RightScale :)
>> >
>> > Oooops, I'll learn to pay attention to signatures :)
>> >
>> >
>> >>
>> >>
>> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>> >>
>> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>> >>>> RightScale's management dashboard"?
>> >>>>
>> >>>> There are some significant differences to the configuration when
>> >>>> using RightScale vs. bare bones AWS; and given this is on my
>> >>>> employer's time I need to keep it relevant...
>> >>>
>> >>> Chris,
>> >>> I know can be more work, but the best would be to let know the
>> >>> differences too, and how it would be using bare bone AWS. That's also
>> >>> good for your employer, if he decide to switch back from RightScale...
>> >>> :)
>> >>>
>> >>> -giovanni
>> >>>
>> >>>
>> >>>>
>> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>> >>>> a public AMI (you don't need to be a RightScale customer to use it)
>> >>>>
>> >>>> Cheers, Chris.
>> >>>>
>> >>>> -----Original Message-----
>> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org
>> >>>> ] On Behalf Of Russell.Mosemann at cune.org
>> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
>> >>>> To: freeswitch-users at lists.freeswitch.org
>> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>> >>>>
>> >>>> Chris Fowler <chris at fowler.cc> said:
>> >>>>
>> >>>>> There were no special tricks; you do need to modify/override the
>> >>>>> following with the box's Elastic IP (EIP).
>> >>>>
>> >>>> This would be helpful to have in the wiki along with any other tips
>> >>>> for
>> >>>> your virtual environment. :-)
>> >>>>
>> >>>> --
>> >>>> Russell Mosemann
>> >>>>
>> >>>>
>> >>>>
>> >>>> ________________________________________________________
>> >>>> Concordia University, Nebraska
>> >>>> See http://www.cune.edu/ for the latest news and events!
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Sincerely,
>> >>>
>> >>> Giovanni Maruzzelli
>> >>> Cell : +39-347-2665618
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:14:36 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:14:36 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] Call transfer using
	eventsockets with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
References: <Acq8U5J/iQUC0nAxSbWQSsxFzefS/Q==>
	<3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072014h4368cb2x1a02c0fb5953504d@mail.gmail.com>

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

Regards,
Jo?o Mesquita


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

>  Hi,
>
>
>
> Is there a way to transfer call with FSComm. I tried different ways using
> eventsockets but failed.
>
>
>
> Tried the scenario as follows:
>
> Received a call from 1001 to FSComm (registered with 1002) and is answered
> (Unique-id is 784dd690-6d0a-47de-b5f4-b923264581a5). Made a call from FSComm
> to extension 1003 and is answered (Unique-id is
> ec7a3bdd-c265-41fb-8cef-b897d2e8bf62). Now I want to transfer this call to
> 1001 and FSComm should be out of loop.
>
>
>
> a.       Tried with command ?api uuid_transfer
> 784dd690-6d0a-47de-b5f4-b923264581a5 ?both park inline? but failed. With
> this call at 1001 is dropped and call between FSComm and 1003 is still
> there.
>
> b.      Tried with command ?api uuid_bridge
> 784dd690-6d0a-47de-b5f4-b923264581a5 ec7a3bdd-c265-41fb-8cef-b897d2e8bf62?.
> With this all calls are dropped.
>
>
>
> Could someone let me know if there is any procedure for call transfer
> (using event sockets).
>
>
>
> Thanks & Regards
>
> Suneel
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
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> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
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> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:15:19 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:15:19 -0300
Subject: [Freeswitch-users] FScomm
In-Reply-To: <13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
	<D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
	<ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>
	<13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
Message-ID: <ebbc2d551003072015t7537eae4g6564c1286fb1fd3b@mail.gmail.com>

If I can access your system over SSH I can probably figure this out. Is this
possible?

Jo?o Mesquita


On Wed, Mar 3, 2010 at 3:26 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi Joao,
>
> thanks for your answer.
> I have a standard RPM QT last update for Fedora10 64bits.
> so I really don't know what path FScomm needs
>
> Regards
>
> Franck
>
> ----- Original Message -----
> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 03, 2010 1:00 AM
> *Subject:* Re: [Freeswitch-users] FScomm
>
> This is definitely a problem with your Qt installation. You need to be able
> to find the libs to link it, otherwise, unresolved symbols everywhere!
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:
>
>>
>> ----- Original Message -----
>> From: "Michael Jerris" <mike at jerris.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, February 24, 2010 3:40 AM
>>  Subject: Re: [Freeswitch-users] FScomm
>>
>>
>>
>> On Feb 23, 2010, at 6:18 PM, Madovsky wrote:
>>
>> >
>> > ----- Original Message -----
>> > From: Jeff Lenk
>> > To: freeswitch-users at lists.freeswitch.org
>> > Sent: Tuesday, February 23, 2010 5:05 PM
>> > Subject: Re: [Freeswitch-users] FScomm
>> >
>> >
>> > http://wiki.freeswitch.org/wiki/FSComm#Linux
>> >
>> > you must run those from the FSComm directory
>> >
>>
>> ?.
>>
>> >
>> > It's what I did,
>> > but from FS trunk, inside fscomm directory,
>> > there s only
>> >
>> > account.cpp  conf                fshost.h        mainwindow.ui
>> > resources.qrc
>> > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
>> > call.cpp     FSComm.pro          mainwindow.cpp  preferences
>> > call.h       fshost.cpp          mainwindow.h    resources
>> >
>>
>>
>> Read those installation instructions again and do them step by step, you
>> skipped one.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> Ok now I have
>>
>> [root at node250 fscomm]# qmake
>> WARNING: Found potential symbol conflict of mainwindow.cpp
>> (mainwindow.cpp)
>> in SOURCES
>> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
>> HEADERS
>> WARNING: Found potential symbol conflict of prefdialog.cpp
>> (preferences/prefdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of prefdialog.h
>> (preferences/prefdialog.h) in HEADERS
>> WARNING: Found potential symbol conflict of accountdialog.cpp
>> (preferences/accountdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of accountdialog.h
>> (preferences/accountdialog.h) in HEADERS
>>
>> [root at node250 fscomm]# qmake
>> WARNING: Found potential symbol conflict of mainwindow.cpp
>> (mainwindow.cpp)
>> in SOURCES
>> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
>> HEADERS
>> WARNING: Found potential symbol conflict of prefdialog.cpp
>> (preferences/prefdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of prefdialog.h
>> (preferences/prefdialog.h) in HEADERS
>> WARNING: Found potential symbol conflict of accountdialog.cpp
>> (preferences/accountdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of accountdialog.h
>> (preferences/accountdialog.h) in HEADERS
>> [root at node250 fscomm]# make
>> Makefile:278: warning: overriding commands for target `prefdialog.o'
>> Makefile:215: warning: ignoring old commands for target `prefdialog.o'
>> Makefile:285: warning: overriding commands for target `accountdialog.o'
>> Makefile:234: warning: ignoring old commands for target `accountdialog.o'
>> Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
>> Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
>> Makefile:323: warning: overriding commands for target
>> `moc_accountdialog.o'
>> Makefile:307: warning: ignoring old commands for target
>> `moc_accountdialog.o'
>> Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
>> Makefile:326: warning: ignoring old commands for target
>> `moc_mainwindow.cpp'
>> Makefile:350: warning: overriding commands for target
>> `preferences/moc_prefdialog.cpp'
>> Makefile:332: warning: ignoring old commands for target
>> `preferences/moc_prefdialog.cpp'
>> Makefile:353: warning: overriding commands for target
>> `preferences/moc_accountdialog.cpp'
>> Makefile:341: warning: ignoring old commands for target
>> `preferences/moc_accountdialog.cpp'
>> g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
>> -fexceptions
>>  -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic
>>  -DQT_NO_DEBUG
>>  -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT
>> -I/usr/lib64/qt-3.3/mkspecs/default
>>  -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src
>> -I/usr/lib64/qt-3.3/include
>>  -o main.o main.cpp
>> main.cpp:31:25: error: QSplashScreen: No such file or directory
>> In file included from main.cpp:32:
>> mainwindow.h:34:23: error: QMainWindow: No such file or directory
>> mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
>> mainwindow.h:36:25: error: QSignalMapper: No such file or directory
>> mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
>> In file included from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./fshost.h:32:19: error: QThread: No such file or directory
>> ./fshost.h:33:17: error: QHash: No such file or directory
>> ./fshost.h:34:26: error: QSharedPointer: No such file or directory
>> In file included from ./fshost.h:36,
>>                 from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./call.h:32:18: error: QtCore: No such file or directory
>> ./call.h:33:19: error: QString: No such file or directory
>> In file included from mainwindow.h:42,
>>                 from main.cpp:32:
>> preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
>> preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
>> directory
>> preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
>> In file included from ./fshost.h:37,
>>                 from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./account.h:18: error: expected constructor, destructor, or type
>> conversion
>> before ?static?
>> In file included from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
>> /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
>> QThread?
>> ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:46: error: expected ?;? before ?<? token
>> ./fshost.h:47: error: expected `;' before ?QSharedPointer?
>> ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:47: error: expected ?;? before ?<? token
>> ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:48: error: template argument 1 is invalid
>> ./fshost.h:48: error: expected unqualified-id before ?>? token
>> ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:49: error: expected ?;? before ?<? token
>> ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:50: error: expected ?;? before ?<? token
>> ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:51: error: expected ?;? before ?<? token
>> ./fshost.h:52: error: ?QSharedPointer? has not been declared
>> ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:60: error: ?QSharedPointer? has not been declared
>> ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:61: error: ?QSharedPointer? has not been declared
>> ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:62: error: ?QSharedPointer? has not been declared
>> ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:63: error: ?QSharedPointer? has not been declared
>> ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:64: error: ?QSharedPointer? has not been declared
>> ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:65: error: ?QSharedPointer? has not been declared
>> ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:66: error: ?QSharedPointer? has not been declared
>> ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:67: error: ?QSharedPointer? has not been declared
>> ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:71: error: ?QSharedPointer? has not been declared
>> ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:78: error: template argument 2 is invalid
>> ./fshost.h:78: error: expected unqualified-id before ?>? token
>> ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:79: error: template argument 2 is invalid
>> ./fshost.h:79: error: expected unqualified-id before ?>? token
>> ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
>> In file included from mainwindow.h:42,
>>                 from main.cpp:32:
>> preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
>> QDialog?
>> /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
>> QDialog?
>> preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
>> ?QSettings? with no type
>> preferences/prefdialog.h:31: error: expected ?;? before ?*? token
>> In file included from main.cpp:32:
>> mainwindow.h:48: error: expected class-name before ?{? token
>> mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
>> mainwindow.h:71: error: ?QSharedPointer? has not been declared
>> mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:72: error: ?QSharedPointer? has not been declared
>> mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:73: error: ?QSharedPointer? has not been declared
>> mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:74: error: ?QSharedPointer? has not been declared
>> mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:75: error: ?QSharedPointer? has not been declared
>> mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:78: error: ?QSharedPointer? has not been declared
>> mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:79: error: ?QSharedPointer? has not been declared
>> mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:80: error: ?QSharedPointer? has not been declared
>> mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
>> mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
>> mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper?
>> with
>> no type
>> mainwindow.h:86: error: expected ?;? before ?*? token
>> mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
>> with no type
>> mainwindow.h:88: error: expected ?;? before ?*? token
>> main.cpp: In function ?int main(int, char**)?:
>> main.cpp:41: error: variable ?QPixmap image? has initializer but
>> incomplete
>> type
>> main.cpp:42: error: ?QSplashScreen? was not declared in this scope
>> main.cpp:42: error: ?splash? was not declared in this scope
>> main.cpp:42: error: expected type-specifier before ?QSplashScreen?
>> main.cpp:42: error: expected `;' before ?QSplashScreen?
>> main.cpp:48: error: no matching function for call to
>> ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
>> /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
>> QObject::connect(const QObject*, const char*, const QObject*, const char*,
>> Qt::ConnectionType)
>> /usr/include/QtCore/qobject.h:307: note:                 bool
>> QObject::connect(const QObject*, const char*, const char*,
>> Qt::ConnectionType) const
>> main.cpp:49: error: ?class FSHost? has no member named ?start?
>> main.cpp:41: warning: unused variable ?image?
>> ./fshost.h: At global scope:
>> ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)?
>> defined
>> but not used
>> make: *** [main.o] Error 1
>>
>>
>> Any idea ?
>>
>> Thx
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:18:59 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:18:59 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] FSComm basic issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
References: <Acq3BRPNPSeAcwujTl+n9yQujbXGcA==>
	<3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>

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

Inline...

JM

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

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

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

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

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

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


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


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

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

>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *************************************************************************
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> are addressed. If you have received this email in error please notify
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>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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>
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From freeswitch.org at todandlorna.com  Sun Mar  7 21:28:46 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Sun, 07 Mar 2010 22:28:46 -0700
Subject: [Freeswitch-users] Conference Recording
In-Reply-To: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
References: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
Message-ID: <4B948B0E.8050909@todandlorna.com>

For the whole thing?  I know parts were recorded by different people.  
I'm not sure about the whole thing.

-Tod Hansmann

On 3/7/2010 5:28 PM, Josh Rivers wrote:
> On last week's conference call, someone was recording, did that 
> recording get put online somewhere?
>
> Josh
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From infos at madovsky.org  Sun Mar  7 22:20:01 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 01:20:01 -0500
Subject: [Freeswitch-users] registration in a DB shared mode
Message-ID: <AF56DAE1642E48AB848D8F5761DAE125@MOBILEE1705>

I set 2 freeswitch on 2 nodes with a shared DB,
set all configs for shared db with ODBC postgresql,
but if I call an extension registered from the another node 
voicemail answer. I missed to set something somewhere,
if anyone can help it would be great

Regards

Franck
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From jingwei.yang at gmail.com  Mon Mar  8 00:43:01 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 8 Mar 2010 16:43:01 +0800
Subject: [Freeswitch-users] How to set background music for application park
Message-ID: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>

Hello,

I'm using this command to put each party of a call to hold:

uuid_transfer <uuid> -both 'park' inline

It works perfectly except none of them can hear the background music. May I
know how to set the music in this case? And is it possible to set different
music for different parties?

Thanks,
-Jingwei
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From gavin.henry at gmail.com  Mon Mar  8 01:34:34 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 09:34:34 +0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
Message-ID: <13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>

We use a 256mb xen vm for customers hosted pbx platforms for up to 20
users. Everything is fine re conferencing and voicemails etc. We also
use the -vm flag and this is driven by FusionPBX

Each customer gets their own vm so not much load.

Thanks,

Gavin.

On 08/03/2010, Rob Forman <rob4manhere at gmail.com> wrote:
> Nice to see so many people using it in so many different ways.  For those
> using ec2 or xen, have you noticed this in your logs?
>
> 2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating timer,
> please wait...
> 2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer set to higher than 1 kHz? You may experience
> audio problems.
>
>
> "You may experience audio problems."  Seems pretty clear.  Can that just be
> ignored?
>
> Rob
>
>
> On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
>> wrote:
>
>> I used to do tests on conferencing using linode.com, which is Xen VMs
>> without issues, although there were not much load on it.
>>
>> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
>>> pretty well. Of course no much load.
>>>
>>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>>> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
>>> wrote:
>>> >>  >  Cheers, Chris.
>>> >>  >  --
>>> >>  >  RightScale, Inc.
>>> >>
>>> >> I doubt they'll be switching any time soon.  I think his employer IS
>>> >> RightScale :)
>>> >
>>> > Oooops, I'll learn to pay attention to signatures :)
>>> >
>>> >
>>> >>
>>> >>
>>> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>> >>
>>> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>>> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>> >>>> RightScale's management dashboard"?
>>> >>>>
>>> >>>> There are some significant differences to the configuration when
>>> >>>> using RightScale vs. bare bones AWS; and given this is on my
>>> >>>> employer's time I need to keep it relevant...
>>> >>>
>>> >>> Chris,
>>> >>> I know can be more work, but the best would be to let know the
>>> >>> differences too, and how it would be using bare bone AWS. That's also
>>> >>> good for your employer, if he decide to switch back from
>>> >>> RightScale...
>>> >>> :)
>>> >>>
>>> >>> -giovanni
>>> >>>
>>> >>>
>>> >>>>
>>> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>> >>>> a public AMI (you don't need to be a RightScale customer to use it)
>>> >>>>
>>> >>>> Cheers, Chris.
>>> >>>>
>>> >>>> -----Original Message-----
>>> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org
>>> >>>> ] On Behalf Of Russell.Mosemann at cune.org
>>> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>> >>>> To: freeswitch-users at lists.freeswitch.org
>>> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>> >>>>
>>> >>>> Chris Fowler <chris at fowler.cc> said:
>>> >>>>
>>> >>>>> There were no special tricks; you do need to modify/override the
>>> >>>>> following with the box's Elastic IP (EIP).
>>> >>>>
>>> >>>> This would be helpful to have in the wiki along with any other tips
>>> >>>> for
>>> >>>> your virtual environment. :-)
>>> >>>>
>>> >>>> --
>>> >>>> Russell Mosemann
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> ________________________________________________________
>>> >>>> Concordia University, Nebraska
>>> >>>> See http://www.cune.edu/ for the latest news and events!
>>> >>>>
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> --
>>> >>> Sincerely,
>>> >>>
>>> >>> Giovanni Maruzzelli
>>> >>> Cell : +39-347-2665618
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Sincerely,
>>> >
>>> > Giovanni Maruzzelli
>>> > Cell : +39-347-2665618
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jaybinks at gmail.com  Mon Mar  8 03:30:48 2010
From: jaybinks at gmail.com (Jay Binks)
Date: Mon, 8 Mar 2010 21:30:48 +1000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
	<13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
Message-ID: <70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>

The qn here is do you bypass media or do you do complete topology  
hiding ??

Sip only is kind of a given ,  it gets harder if the virtualized pbx  
does media.

J



On 08/03/2010, at 19:34, Gavin Henry <gavin.henry at gmail.com> wrote:

> We use a 256mb xen vm for customers hosted pbx platforms for up to 20
> users. Everything is fine re conferencing and voicemails etc. We also
> use the -vm flag and this is driven by FusionPBX
>
> Each customer gets their own vm so not much load.
>
> Thanks,
>
> Gavin.
>
> On 08/03/2010, Rob Forman <rob4manhere at gmail.com> wrote:
>> Nice to see so many people using it in so many different ways.  For  
>> those
>> using ec2 or xen, have you noticed this in your logs?
>>
>> 2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating  
>> timer,
>> please wait...
>> 2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer  
>> resolution of
>> 4000 microseconds detected!
>> Do you have your kernel timer set to higher than 1 kHz? You may  
>> experience
>> audio problems.
>>
>>
>> "You may experience audio problems."  Seems pretty clear.  Can that  
>> just be
>> ignored?
>>
>> Rob
>>
>>
>> On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
>>> wrote:
>>
>>> I used to do tests on conferencing using linode.com, which is Xen  
>>> VMs
>>> without issues, although there were not much load on it.
>>>
>>> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com>  
>>> wrote:
>>>
>>>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit),  
>>>> seems
>>>> pretty well. Of course no much load.
>>>>
>>>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>>>>> On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman  
>>>>> <rob4manhere at gmail.com>
>>>> wrote:
>>>>>>> Cheers, Chris.
>>>>>>> --
>>>>>>> RightScale, Inc.
>>>>>>
>>>>>> I doubt they'll be switching any time soon.  I think his  
>>>>>> employer IS
>>>>>> RightScale :)
>>>>>
>>>>> Oooops, I'll learn to pay attention to signatures :)
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>>>>>
>>>>>>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc>  
>>>>>>> wrote:
>>>>>>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>>>>>>>> have an objection if I create a "How to use FreeSWITCH on EC2  
>>>>>>>> with
>>>>>>>> RightScale's management dashboard"?
>>>>>>>>
>>>>>>>> There are some significant differences to the configuration  
>>>>>>>> when
>>>>>>>> using RightScale vs. bare bones AWS; and given this is on my
>>>>>>>> employer's time I need to keep it relevant...
>>>>>>>
>>>>>>> Chris,
>>>>>>> I know can be more work, but the best would be to let know the
>>>>>>> differences too, and how it would be using bare bone AWS.  
>>>>>>> That's also
>>>>>>> good for your employer, if he decide to switch back from
>>>>>>> RightScale...
>>>>>>> :)
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10";  
>>>>>>>> this is
>>>>>>>> a public AMI (you don't need to be a RightScale customer to  
>>>>>>>> use it)
>>>>>>>>
>>>>>>>> Cheers, Chris.
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org
>>>>>>>> ] On Behalf Of Russell.Mosemann at cune.org
>>>>>>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>>>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>>>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>>>>>>
>>>>>>>> Chris Fowler <chris at fowler.cc> said:
>>>>>>>>
>>>>>>>>> There were no special tricks; you do need to modify/override  
>>>>>>>>> the
>>>>>>>>> following with the box's Elastic IP (EIP).
>>>>>>>>
>>>>>>>> This would be helpful to have in the wiki along with any  
>>>>>>>> other tips
>>>>>>>> for
>>>>>>>> your virtual environment. :-)
>>>>>>>>
>>>>>>>> --
>>>>>>>> Russell Mosemann
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ________________________________________________________
>>>>>>>> Concordia University, Nebraska
>>>>>>>> See http://www.cune.edu/ for the latest news and events!
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> -- 
> Sent from my mobile device
>
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From jbrucehopkins at gmail.com  Mon Mar  8 04:56:12 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 12:56:12 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
Message-ID: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>

Hi,

I wonder if some kind soul might be able please to help me by pointing out
what I am doing wrong with my doublenat configuration.

I have incoming calls from WAN --> LAN working fine, but when I try to call
from LAN -->WAN, the Freeswitch CLI says:


"[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type
[error] cause: [USER_NOT_REGISTERED]"


The setup is essentially:


phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT
(public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)


Phone_1 is able to register with Freeswitch using a "doublenat" sip profile
I set up using port 5090, created from the default external profile with the
port changed. phone_1 can initiate calls to phone_2 fine with no problems
with audio - everything works just fine in that direction.

However when I try to call from phone_2 --> phone_1 I get the message "The
person at extension 1002 is not available.  Record your message at the
tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not
registered.

In trying to solve this I have run a Wireshark trace onthe PC running the
phone_1 softphone to check that the SIP registration seems to go as I
understand it should.  the dialog is as follows:

phone_1 --> FS             request: REGISTER
FS         -->  phone_1    status: 401 Unauthorised
phone_1 --> FS             request: REGISTER
FS         --> phone_1     status: 200 OK
phone_1 --> FS             request: SUBSCRIBE
FS         --> phone_1     status: 405 Method Not Allowed


There does appear to be a difference here to what happens when phone_2 (on
the same LAN as Freeswitch) registers.  With phone_2 I see:

phone_2 --> FS             request: REGISTER
FS         --> phone_2     status: 401 Unauthorised
phone_2 --> FS             request: REGISTER
FS         --> phone_2     status  200 OK
phone_2 --> FS             request: SUBSCRIBE
FS         --> phone_2     status: 202 Accepted
FS         --> phone_2     request: NOTIFY
phone_2 --> FS             status: 200 OK



More configuration details:

In conf/sip_profiles/doublenat.xml I have not uncommented <param
name="force-register-domain" value="$$(domain)"/>

In conf/sip_profiles/doublenat.xml I have tried setting the context to each
of public and default.  This does not seem to make a difference.

In conf/directory/doublenat.xml I have set the domain name to be the domain
name registerd for Fs's external IP with DNS, and used by phone_1.

In conf/directory/doublenat.xml I have tried both of the following 9I  don't
know if this is relevant)

1. Firstly with phone_1 registering to an extension in the default
configuration, held in conf/directory/default/:

<groups>
    <group name="default">
       <users>
          <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
       </users>
    </group>

2.  Then I tried moving the .xml file for phone_1's extension to
conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:

<groups>
    <group name="doublenat">
       <users>
          <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
       </users>
     </group>

Unfortunately this does not make a difference to the issue though.

If anybody had the time to tell me what I am doing wrong I would be hugely
grateful !

Many thanks in advance
Bruce
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From jbrucehopkins at gmail.com  Mon Mar  8 05:25:35 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 13:25:35 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
Message-ID: <d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>

Hi Anthony,

Many, many apologies to take so long to reply - I failed to notice your
response.  Sorry !

It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
then does not give a warning.

Best wishes
Bruce

On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> we are still wary about 5.3 due to bugs reported in libc.
> What number does it say it detected for the gap?
>
>
> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>
>> OK - I've realised I do get the same warning with CentOS 5.3, it just goes
>> past more quickly so I didn't see it.  Maybe it is just the hardware ....
>>
>>
>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I wonder if anyone would be able to advise please:
>>>
>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
>>> start FreeSWITCH that
>>>
>>> "Abnormally large timer gap detected"
>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>> experience audio problems".
>>>
>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>> to Centos5.4 was a yum update on the 5.3 build.
>>>
>>> I guess the warning comes from here:
>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>
>>> This is all on pretty low spec hardware - a couple of different Dell
>>> optiplex p4's I use for testing.
>>>
>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>> use such crummy hardware?!
>>>
>>> Many thanks in advance
>>> Bruce
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shaheryarkh at googlemail.com  Mon Mar  8 05:38:42 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 18:38:42 +0500
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
Message-ID: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>

Hi,

I am trying to enable ZRTP support in FS, following the instructions given
at http://wiki.freeswitch.org/wiki/ZRTP.

Upon executing Step 4, i get following error, afterwords it starts some
configure script.

root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
tar: Error is not recoverable: exiting now
tar: Child returned status 2
tar: Error exit delayed from previous errors
cd: 3: can't cd to libzrtp-0.81.514
build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such
file
cd: 5: can't cd to projects/gnu/
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... /bin/mkdir -p
checking for gawk... no
checking for mawk... mawk
checking whether make sets $(MAKE)... yes
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
....

I have no idea, how to fix this. Please help.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Mon Mar  8 06:26:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:26:32 -0600
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
In-Reply-To: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
References: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
Message-ID: <920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>

step 1. get the latest libzrtp from their website.  Step 2 you don't need the patches now. Step 3. compile and install the lib with --enable-pic

/b


On Mar 8, 2010, at 7:38 AM, Muhammad Shahzad wrote:

> Hi,
> 
> I am trying to enable ZRTP support in FS, following the instructions given at http://wiki.freeswitch.org/wiki/ZRTP.
> 
> Upon executing Step 4, i get following error, afterwords it starts some configure script.
> 
> root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
> tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
> tar: Error is not recoverable: exiting now
> tar: Child returned status 2
> tar: Error exit delayed from previous errors
> cd: 3: can't cd to libzrtp-0.81.514
> build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such file
> cd: 5: can't cd to projects/gnu/
> checking for a BSD-compatible install... /usr/bin/install -c
> checking whether build environment is sane... yes
> checking for a thread-safe mkdir -p... /bin/mkdir -p
> checking for gawk... no
> checking for mawk... mawk
> checking whether make sets $(MAKE)... yes
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> ....
> 
> I have no idea, how to fix this. Please help.
> 
> Thank you.
> 
> 
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Mar  8 06:28:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:28:05 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
Message-ID: <4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>

Step 1. Revert everything back to SVN trunk.
Step 2. Enable UPNP/NATPMP on both routers.
Step 3. It should just work.  The doublenat profile is NO LONGER NEEDED.

If you can't enable UPNP/NATPMP then by all means static map the ports correctly or put the FreeSWITCH box in the DMZ on both ends.  It should just work.

/b

On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:

> Hi,
>  
> I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration.
>  
> I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says:
>  
>  
> "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
>  
>  
> The setup is essentially:
>  
>  
> phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
>  
>  
> Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction. 
>  
> However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available.  Record your message at the tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not registered.
>  
> In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should.  the dialog is as follows:
>  
> phone_1 --> FS             request: REGISTER
> FS         -->  phone_1    status: 401 Unauthorised
> phone_1 --> FS             request: REGISTER
> FS         --> phone_1     status: 200 OK
> phone_1 --> FS             request: SUBSCRIBE
> FS         --> phone_1     status: 405 Method Not Allowed
>  
>  
> There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers.  With phone_2 I see:
>  
> phone_2 --> FS             request: REGISTER
> FS         --> phone_2     status: 401 Unauthorised
> phone_2 --> FS             request: REGISTER
> FS         --> phone_2     status  200 OK
> phone_2 --> FS             request: SUBSCRIBE
> FS         --> phone_2     status: 202 Accepted
> FS         --> phone_2     request: NOTIFY
> phone_2 --> FS             status: 200 OK
>  
>  
>  
> More configuration details:
>  
> In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/>
>  
> In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default.  This does not seem to make a difference.
>  
> In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1.
>  
> In conf/directory/doublenat.xml I have tried both of the following 9I  don't know if this is relevant)
>  
> 1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/:
>  
> <groups>
>     <group name="default">
>        <users>
>           <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
>        </users>
>     </group>
>  
> 2.  Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
>  
> <groups>
>     <group name="doublenat">
>        <users>
>           <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
>        </users>
>      </group>
>  
> Unfortunately this does not make a difference to the issue though.
>  
> If anybody had the time to tell me what I am doing wrong I would be hugely grateful !
>  
> Many thanks in advance
> Bruce
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Mar  8 06:30:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:30:09 -0600
Subject: [Freeswitch-users] How to set background music for application
	park
In-Reply-To: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>
References: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>
Message-ID: <75652D86-1C32-4FF7-B9AE-B328EC069BAB@freeswitch.org>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

/b

On Mar 8, 2010, at 2:43 AM, Jingwei Yang wrote:

> Hello,
> 
> I'm using this command to put each party of a call to hold:
> 
> uuid_transfer <uuid> -both 'park' inline
> 
> It works perfectly except none of them can hear the background music. May I know how to set the music in this case? And is it possible to set different music for different parties?
> 
> Thanks,
> -Jingwei
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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

Hello FreeSWITCHers,

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

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

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

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

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From shaheryarkh at googlemail.com  Mon Mar  8 06:53:55 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 19:53:55 +0500
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
In-Reply-To: <920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>
References: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
	<920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>
Message-ID: <b16156851003080653w7db644e8q7cc7277b42a2bf4d@mail.gmail.com>

The file i downloaded from their site is named as zfone-linux.tar. Let me
install that and rebuild FS to see if zrtp support compiles.

Thank you.


On Mon, Mar 8, 2010 at 7:26 PM, Brian West <brian at freeswitch.org> wrote:

> step 1. get the latest libzrtp from their website.  Step 2 you don't need
> the patches now. Step 3. compile and install the lib with --enable-pic
>
> /b
>
>
> On Mar 8, 2010, at 7:38 AM, Muhammad Shahzad wrote:
>
> Hi,
>
> I am trying to enable ZRTP support in FS, following the instructions given
> at http://wiki.freeswitch.org/wiki/ZRTP.
>
> Upon executing Step 4, i get following error, afterwords it starts some
> configure script.
>
> root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
> tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
> tar: Error is not recoverable: exiting now
> tar: Child returned status 2
> tar: Error exit delayed from previous errors
> cd: 3: can't cd to libzrtp-0.81.514
> build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such
> file
> cd: 5: can't cd to projects/gnu/
> checking for a BSD-compatible install... /usr/bin/install -c
> checking whether build environment is sane... yes
> checking for a thread-safe mkdir -p... /bin/mkdir -p
> checking for gawk... no
> checking for mawk... mawk
> checking whether make sets $(MAKE)... yes
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> ....
>
> I have no idea, how to fix this. Please help.
>
> Thank you.
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From jbrucehopkins at gmail.com  Mon Mar  8 07:03:55 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 15:03:55 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
Message-ID: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>

Many thanks for getting back to me with help so quickly Brian.

I was using 1.0.5-20100302-0400 so will update to trunk and do as you
suggest.

thanks again
Bruce

On 8 March 2010 14:28, Brian West <brian at freeswitch.org> wrote:

> Step 1. Revert everything back to SVN trunk.
> Step 2. Enable UPNP/NATPMP on both routers.
> Step 3. It should just work.  The doublenat profile is NO LONGER NEEDED.
>
> If you can't enable UPNP/NATPMP then by all means static map the ports
> correctly or put the FreeSWITCH box in the DMZ on both ends.  It should just
> work.
>
> /b
>
> On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:
>
> > Hi,
> >
> > I wonder if some kind soul might be able please to help me by pointing
> out what I am doing wrong with my doublenat configuration.
> >
> > I have incoming calls from WAN --> LAN working fine, but when I try to
> call from LAN -->WAN, the Freeswitch CLI says:
> >
> >
> > "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type
> [error] cause: [USER_NOT_REGISTERED]"
> >
> >
> > The setup is essentially:
> >
> >
> > phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT
> (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
> >
> >
> > Phone_1 is able to register with Freeswitch using a "doublenat" sip
> profile I set up using port 5090, created from the default external profile
> with the port changed. phone_1 can initiate calls to phone_2 fine with no
> problems with audio - everything works just fine in that direction.
> >
> > However when I try to call from phone_2 --> phone_1 I get the message
> "The person at extension 1002 is not available.  Record your message at the
> tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not
> registered.
> >
> > In trying to solve this I have run a Wireshark trace onthe PC running the
> phone_1 softphone to check that the SIP registration seems to go as I
> understand it should.  the dialog is as follows:
> >
> > phone_1 --> FS             request: REGISTER
> > FS         -->  phone_1    status: 401 Unauthorised
> > phone_1 --> FS             request: REGISTER
> > FS         --> phone_1     status: 200 OK
> > phone_1 --> FS             request: SUBSCRIBE
> > FS         --> phone_1     status: 405 Method Not Allowed
> >
> >
> > There does appear to be a difference here to what happens when phone_2
> (on the same LAN as Freeswitch) registers.  With phone_2 I see:
> >
> > phone_2 --> FS             request: REGISTER
> > FS         --> phone_2     status: 401 Unauthorised
> > phone_2 --> FS             request: REGISTER
> > FS         --> phone_2     status  200 OK
> > phone_2 --> FS             request: SUBSCRIBE
> > FS         --> phone_2     status: 202 Accepted
> > FS         --> phone_2     request: NOTIFY
> > phone_2 --> FS             status: 200 OK
> >
> >
> >
> > More configuration details:
> >
> > In conf/sip_profiles/doublenat.xml I have not uncommented <param
> name="force-register-domain" value="$$(domain)"/>
> >
> > In conf/sip_profiles/doublenat.xml I have tried setting the context to
> each of public and default.  This does not seem to make a difference.
> >
> > In conf/directory/doublenat.xml I have set the domain name to be the
> domain name registerd for Fs's external IP with DNS, and used by phone_1.
> >
> > In conf/directory/doublenat.xml I have tried both of the following 9I
>  don't know if this is relevant)
> >
> > 1. Firstly with phone_1 registering to an extension in the default
> configuration, held in conf/directory/default/:
> >
> > <groups>
> >     <group name="default">
> >        <users>
> >           <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
> >        </users>
> >     </group>
> >
> > 2.  Then I tried moving the .xml file for phone_1's extension to
> conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
> >
> > <groups>
> >     <group name="doublenat">
> >        <users>
> >           <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
> >        </users>
> >      </group>
> >
> > Unfortunately this does not make a difference to the issue though.
> >
> > If anybody had the time to tell me what I am doing wrong I would be
> hugely grateful !
> >
> > Many thanks in advance
> > Bruce
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Mon Mar  8 07:08:34 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 8 Mar 2010 20:38:34 +0530
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
Message-ID: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>

Hi,

can any body tried to connect cisco router from Freeswitch, please help me.

-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Mar  8 07:15:01 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:15:01 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
Message-ID: <C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>

Well sounds like you need to sync in your configs too.

/b

On Mar 8, 2010, at 9:03 AM, Bruce Hopkins wrote:

> Many thanks for getting back to me with help so quickly Brian.
> 
> I was using 1.0.5-20100302-0400 so will update to trunk and do as you suggest.
> 
> thanks again
> Bruce



From shaheryarkh at googlemail.com  Mon Mar  8 07:16:34 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 20:16:34 +0500
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <b16156851003080716l6e6d5b0s47bc57724ab14eda@mail.gmail.com>

I have CISCO AS 5350 running with Freeswitch in production environment.
Seems to work perfect.

Thank you.


On Mon, Mar 8, 2010 at 8:08 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From yehavi.bourvine at gmail.com  Mon Mar  8 07:18:45 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 8 Mar 2010 17:18:45 +0200
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <d361ca7b1003080718l6aaf9db7t6f72707ac17cded9@mail.gmail.com>

What help do you need? What Cisco do you have?

                __Yehavi:

 2010/3/8 srinivasula reddy <srinivas.ksvreddy at gmail.com>

>  Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Mon Mar  8 07:23:35 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 8 Mar 2010 20:53:35 +0530
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>

HI,

how do i configure freeswitch gateway to communicate to cisco router, i am
using cisco2921.

Thanks
Srinivas
On Mon, Mar 8, 2010 at 8:38 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Mar  8 07:33:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:33:34 -0600
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
Message-ID: <9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>

You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.

/b

On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:

> 
> HI,
>  
> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
>  
> Thanks
> Srinivas



From kristian.kielhofner at gmail.com  Mon Mar  8 07:40:50 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Mar 2010 10:40:50 -0500
Subject: [Freeswitch-users] Attrafax
Message-ID: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>

Disclaimers:

- I haven't looked at it in detail (at all, really)
- It is GPL2 (although perhaps Zoa and co can budge on this)

http://www.venturevoip.com/news.php?rssid=2364

Any thoughts?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From vizentini at hotmail.com  Mon Mar  8 07:42:43 2010
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Mon, 8 Mar 2010 15:42:43 +0000
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>,
	<4B9409FF.5060608@gmail.com>,
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>,
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>,
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>,
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>,
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>,
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
Message-ID: <SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>



Hi
"As per FS wiki we can only return one XML extension upon FS request. "
Could you please tell me where did you find that info? I do return several extensions by request and FS works as expect (Am I wrong doing that?)
"Now we need to return a set of extensions with same number
condition and context but with different second condition so that FS
can pick up the right one. 
Is it possible at all? And if yes how response XML with multiple extensions should look like?"
Yes, It seems to be possible (Am I wrong?) but notice that you are giving the  same extension name for all your extensions:
 <extension name="test9">

I think this can mislead dialplan_hunt in mod_dialplan_xml...Give each extension a unique name, and FS will parse/regex all them for you... 

All the best,Paulo
> From: brian at freeswitch.org
> Date: Sun, 7 Mar 2010 20:17:19 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
> 
> Sounds like you zigged where you should have zagged.
> 
> /b
> 
> On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> 
> > :-))
> >  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> > So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From brian at freeswitch.org  Mon Mar  8 07:47:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:47:44 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
Message-ID: <007EE533-8154-4B00-AEFD-3857E684E9EB@freeswitch.org>

We have SpanDSP and its LGPL.  

/b

On Mar 8, 2010, at 9:40 AM, Kristian Kielhofner wrote:

> Disclaimers:
> 
> - I haven't looked at it in detail (at all, really)
> - It is GPL2 (although perhaps Zoa and co can budge on this)
> 
> http://www.venturevoip.com/news.php?rssid=2364
> 
> Any thoughts?
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Mon Mar  8 07:57:23 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 08 Mar 2010 23:57:23 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
Message-ID: <4B951E63.6060200@coppice.org>

On 03/08/2010 11:40 PM, Kristian Kielhofner wrote:
> Disclaimers:
>
> - I haven't looked at it in detail (at all, really)
> - It is GPL2 (although perhaps Zoa and co can budge on this)
>
> http://www.venturevoip.com/news.php?rssid=2364
>
> Any thoughts?
>    
We already have a better FAX engine than that. What we struggle to find 
is the time to properly integrate it with FS.

Steve



From m.sobkow at marketelsystems.com  Mon Mar  8 08:22:34 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Mar 2010 10:22:34 -0600
Subject: [Freeswitch-users] Eavesdrop
Message-ID: <4B95244A.8030808@marketelsystems.com>

What module contains the "eavesdrop" command?  I'm getting an error 
command not found from Erlang when I try to do a pbx:api( eavesdrop, 
ExtensionUUID ).

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



From kristian.kielhofner at gmail.com  Mon Mar  8 08:24:47 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Mar 2010 11:24:47 -0500
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4B951E63.6060200@coppice.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
Message-ID: <4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>

On Mon, Mar 8, 2010 at 10:57 AM, Steve Underwood <steveu at coppice.org> wrote:
> We already have a better FAX engine than that. What we struggle to find
> is the time to properly integrate it with FS.
>
> Steve

  Thank you, that's what I was expecting.

  Without going into grueling detail it isn't clear why T.38 gateway
functionality (for example) is missing in FreeSWITCH.  I know SpanDSP
is available and most likely has this ability.  I just wasn't sure why
it's missing in FS.  This is also a mere curiosity for me; I don't
have much interest in fax functionality and I don't follow it too
closely.

  Thanks again for the response and explanation.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From rupa at rupa.com  Mon Mar  8 08:55:05 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 8 Mar 2010 10:55:05 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B95244A.8030808@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
Message-ID: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>

application,eavesdrop,mod_dptools,/opt/freeswitch/mod/mod_dptools.so

It is an app, not an API.

On Mon, Mar 8, 2010 at 10:22 AM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> What module contains the "eavesdrop" command?  I'm getting an error
> command not found from Erlang when I try to do a pbx:api( eavesdrop,
> ExtensionUUID ).
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From mike at van.lammeren.net  Mon Mar  8 09:00:37 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 8 Mar 2010 12:00:37 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
	<23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
Message-ID: <5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>

Actually, I don't believe that does work. At least, not on my system. Trying
to access argv[0] just produces a lua error about "concatenating a nil
value."

On Fri, Mar 5, 2010 at 10:28 PM, Seven Du <dujinfang at gmail.com> wrote:

> and also you can get the script name from argv[0] if you name you
> script in different names
>
> 2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
> > In case anyone's interested, I just found something cool. You can pass
> > parameters to the Lua script from the lua.conf.xml file, like so:
> >     <param name="startup-script" value="temp.lua script1"/>
> >     <param name="startup-script" value="temp.lua script2"/>
> >     <param name="startup-script" value="temp.lua script3"/>
> > I can see that being useful.
> > Mike van Lammeren
> >
> > On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net
> >
> > wrote:
> >>
> >> Thanks so much, everyone! Awesome advice, as usual!
> >> Due to a deadline situation, I'm going with the Lua script idea.
> However,
> >> if it gives us any trouble after the looming deadline, then I will
> >> definitely go the C route.
> >> Thanks again!
> >> Mike van Lammeren
> >>
> >> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>>
> >>> Write it in C and use the FreeSWITCH task scheduler API.
> >>>
> >>> /b
> >>>
> >>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
> >>>
> >>> > All the same, I'm still open to suggestions on better ways to achieve
> >>> > my ultimate goals!
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon Mar  8 09:07:03 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 12:07:03 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705><COL117-W5563292C886257AFC56548B9420@phx.gbl><D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705><BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com><D262917F3F4943D49370D0CC015A7582@MOBILEE1705><ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com><13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
	<ebbc2d551003072015t7537eae4g6564c1286fb1fd3b@mail.gmail.com>
Message-ID: <40DCBC09D015499EB5EC5E2B292A3515@MOBILEE1705>

sure, I send to you the connection info now

Regards

  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 07, 2010 11:15 PM
  Subject: Re: [Freeswitch-users] FScomm


  If I can access your system over SSH I can probably figure this out. Is this possible?


  Jo?o Mesquita



  On Wed, Mar 3, 2010 at 3:26 AM, Madovsky <infos at madovsky.org> wrote:

    Hi Joao,

    thanks for your answer.
    I have a standard RPM QT last update for Fedora10 64bits.
    so I really don't know what path FScomm needs

    Regards

    Franck
      ----- Original Message ----- 
      From: Jo?o Mesquita 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 03, 2010 1:00 AM
      Subject: Re: [Freeswitch-users] FScomm


      This is definitely a problem with your Qt installation. You need to be able to find the libs to link it, otherwise, unresolved symbols everywhere! 


      Regards,
      Jo?o Mesquita
      FSComm Developer



      On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:


        ----- Original Message -----
        From: "Michael Jerris" <mike at jerris.com>

        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Wednesday, February 24, 2010 3:40 AM

        Subject: Re: [Freeswitch-users] FScomm



        On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

        >
        > ----- Original Message -----
        > From: Jeff Lenk
        > To: freeswitch-users at lists.freeswitch.org
        > Sent: Tuesday, February 23, 2010 5:05 PM
        > Subject: Re: [Freeswitch-users] FScomm
        >
        >
        > http://wiki.freeswitch.org/wiki/FSComm#Linux
        >
        > you must run those from the FSComm directory
        >

        ?.

        >
        > It's what I did,
        > but from FS trunk, inside fscomm directory,
        > there s only
        >
        > account.cpp  conf                fshost.h        mainwindow.ui
        > resources.qrc
        > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
        > call.cpp     FSComm.pro          mainwindow.cpp  preferences
        > call.h       fshost.cpp          mainwindow.h    resources
        >


        Read those installation instructions again and do them step by step, you
        skipped one.
        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org



        Ok now I have

        [root at node250 fscomm]# qmake
        WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
        in SOURCES
        WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
        HEADERS
        WARNING: Found potential symbol conflict of prefdialog.cpp
        (preferences/prefdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of prefdialog.h
        (preferences/prefdialog.h) in HEADERS
        WARNING: Found potential symbol conflict of accountdialog.cpp
        (preferences/accountdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of accountdialog.h
        (preferences/accountdialog.h) in HEADERS

        [root at node250 fscomm]# qmake
        WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
        in SOURCES
        WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
        HEADERS
        WARNING: Found potential symbol conflict of prefdialog.cpp
        (preferences/prefdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of prefdialog.h
        (preferences/prefdialog.h) in HEADERS
        WARNING: Found potential symbol conflict of accountdialog.cpp
        (preferences/accountdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of accountdialog.h
        (preferences/accountdialog.h) in HEADERS
        [root at node250 fscomm]# make
        Makefile:278: warning: overriding commands for target `prefdialog.o'
        Makefile:215: warning: ignoring old commands for target `prefdialog.o'
        Makefile:285: warning: overriding commands for target `accountdialog.o'
        Makefile:234: warning: ignoring old commands for target `accountdialog.o'
        Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
        Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
        Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
        Makefile:307: warning: ignoring old commands for target
        `moc_accountdialog.o'
        Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
        Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
        Makefile:350: warning: overriding commands for target
        `preferences/moc_prefdialog.cpp'
        Makefile:332: warning: ignoring old commands for target
        `preferences/moc_prefdialog.cpp'
        Makefile:353: warning: overriding commands for target
        `preferences/moc_accountdialog.cpp'
        Makefile:341: warning: ignoring old commands for target
        `preferences/moc_accountdialog.cpp'
        g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
         -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG
         -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default
         -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
         -o main.o main.cpp
        main.cpp:31:25: error: QSplashScreen: No such file or directory
        In file included from main.cpp:32:
        mainwindow.h:34:23: error: QMainWindow: No such file or directory
        mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
        mainwindow.h:36:25: error: QSignalMapper: No such file or directory
        mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
        In file included from mainwindow.h:39,
                        from main.cpp:32:
        ./fshost.h:32:19: error: QThread: No such file or directory
        ./fshost.h:33:17: error: QHash: No such file or directory
        ./fshost.h:34:26: error: QSharedPointer: No such file or directory
        In file included from ./fshost.h:36,
                        from mainwindow.h:39,
                        from main.cpp:32:
        ./call.h:32:18: error: QtCore: No such file or directory
        ./call.h:33:19: error: QString: No such file or directory
        In file included from mainwindow.h:42,
                        from main.cpp:32:
        preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
        preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
        directory
        preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
        In file included from ./fshost.h:37,
                        from mainwindow.h:39,
                        from main.cpp:32:
        ./account.h:18: error: expected constructor, destructor, or type conversion
        before ?static?
        In file included from mainwindow.h:39,
                        from main.cpp:32:
        ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
        /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
        QThread?
        ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:46: error: expected ?;? before ?<? token
        ./fshost.h:47: error: expected `;' before ?QSharedPointer?
        ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:47: error: expected ?;? before ?<? token
        ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:48: error: template argument 1 is invalid
        ./fshost.h:48: error: expected unqualified-id before ?>? token
        ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:49: error: expected ?;? before ?<? token
        ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:50: error: expected ?;? before ?<? token
        ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:51: error: expected ?;? before ?<? token
        ./fshost.h:52: error: ?QSharedPointer? has not been declared
        ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:60: error: ?QSharedPointer? has not been declared
        ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:61: error: ?QSharedPointer? has not been declared
        ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:62: error: ?QSharedPointer? has not been declared
        ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:63: error: ?QSharedPointer? has not been declared
        ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:64: error: ?QSharedPointer? has not been declared
        ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:65: error: ?QSharedPointer? has not been declared
        ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:66: error: ?QSharedPointer? has not been declared
        ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:67: error: ?QSharedPointer? has not been declared
        ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:71: error: ?QSharedPointer? has not been declared
        ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:78: error: template argument 2 is invalid
        ./fshost.h:78: error: expected unqualified-id before ?>? token
        ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:79: error: template argument 2 is invalid
        ./fshost.h:79: error: expected unqualified-id before ?>? token
        ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
        In file included from mainwindow.h:42,
                        from main.cpp:32:
        preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
        QDialog?
        /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
        QDialog?
        preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
        ?QSettings? with no type
        preferences/prefdialog.h:31: error: expected ?;? before ?*? token
        In file included from main.cpp:32:
        mainwindow.h:48: error: expected class-name before ?{? token
        mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
        mainwindow.h:71: error: ?QSharedPointer? has not been declared
        mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:72: error: ?QSharedPointer? has not been declared
        mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:73: error: ?QSharedPointer? has not been declared
        mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:74: error: ?QSharedPointer? has not been declared
        mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:75: error: ?QSharedPointer? has not been declared
        mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:78: error: ?QSharedPointer? has not been declared
        mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:79: error: ?QSharedPointer? has not been declared
        mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:80: error: ?QSharedPointer? has not been declared
        mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
        mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
        mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
        no type
        mainwindow.h:86: error: expected ?;? before ?*? token
        mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
        with no type
        mainwindow.h:88: error: expected ?;? before ?*? token
        main.cpp: In function ?int main(int, char**)?:
        main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
        type
        main.cpp:42: error: ?QSplashScreen? was not declared in this scope
        main.cpp:42: error: ?splash? was not declared in this scope
        main.cpp:42: error: expected type-specifier before ?QSplashScreen?
        main.cpp:42: error: expected `;' before ?QSplashScreen?
        main.cpp:48: error: no matching function for call to
        ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
        /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
        QObject::connect(const QObject*, const char*, const QObject*, const char*,
        Qt::ConnectionType)
        /usr/include/QtCore/qobject.h:307: note:                 bool
        QObject::connect(const QObject*, const char*, const char*,
        Qt::ConnectionType) const
        main.cpp:49: error: ?class FSHost? has no member named ?start?
        main.cpp:41: warning: unused variable ?image?
        ./fshost.h: At global scope:
        ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined
        but not used
        make: *** [main.o] Error 1


        Any idea ?

        Thx




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





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    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jbrucehopkins at gmail.com  Mon Mar  8 09:12:42 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 17:12:42 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
Message-ID: <d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>

Hi again,

Thanks for the help so far.  Unfotrunately I must still be doing something
wrong here as I am still having difficulty, and still have the same problem.

I updated to build 16938 by means of "make current"

I'm not able to use UPNP or NATPMP so changed the port forwarding to use
5080 instead of 5090.

I got rid of the doublenat profile in sip_profiles, though I had to retain
an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml
in order to give freeswitch the dns name of the server as a domain for the
remote softphone to register on.  I left the group name in this entry the
same as inthe default entry, so that the remote phone could register on the
same extension numbers (100, etc) as in the default build.

I still find that, if I initiate a call from the local (on same LAN as
freeswitch) phone to the remote phone, I get the message on the CLI:

[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type
[user] cause: [USER_NOT_REGISTERED]

One possibly unrelated aside, I also found I needed to uncomment <param
name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in
the case of a call initiated by the remote phone being hung up by the local
phone, freeswitch sent the BYE to the private IP of the remote phone, rather
than its public ip - meaning that the remote phone didn't receive the BYE.

Any further ideas where I am going wrong here please?

thanks again in advance
Bruce



On 8 March 2010 15:15, Brian West <brian at freeswitch.org> wrote:

> Well sounds like you need to sync in your configs too.
>
> /b
>
> On Mar 8, 2010, at 9:03 AM, Bruce Hopkins wrote:
>
> > Many thanks for getting back to me with help so quickly Brian.
> >
> > I was using 1.0.5-20100302-0400 so will update to trunk and do as you
> suggest.
> >
> > thanks again
> > Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Sat Mar  6 21:02:42 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sun, 7 Mar 2010 16:02:42 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
Message-ID: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>

Hi,
I am trying to create a mod_managed API application that takes 2 phone
numbers as params.  These numbers should then be used to make 2 calls and to
bridge them. The issue i am having is working out how to place the first
call leg from in mod_managed.  All the methods i have found are for the
second leg and require you to pass in the first call leg.

I know there are easier way to do this out of the managed code but i want it
in the managed code as this is just a proof of concept to prove we can do a
callback like system via managed code to be hooked into other external
systems.
In wiki http://wiki.freeswitch.org/wiki/Session there is this example

s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
while (s.ready()) {
// The call has been answered
}

This seems to be exactly what i am looking for but seems to be missing from
mod_managed. I am hoping someone can tell me how to do something similar in
mod_managed since even if it is not part of the managed code mod_managed is
meant to have the native access as well so i would assume it would be
possible.

Thanks in advance

Chris
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From its.solution at cox.net  Mon Mar  8 07:16:05 2010
From: its.solution at cox.net (ITS Solution)
Date: Mon, 08 Mar 2010 10:16:05 -0500
Subject: [Freeswitch-users] Need help on dingaling module
Message-ID: <C7BA7EE5.3FB3%its.solution@cox.net>

Hi all,

I recently installed FS version 16983 with dingaling module enabled. I am
using the client configuration. My system is Linux Ubuntu 8.04 hardy running
on Xen VM. When I set up a call between a phone and a google talk, the audio
from google talk is all noise. The audio from phone to google talk sounds
good though. Before I did the upgrade, I was running on version  16892. In
that release, I can hear audio on both paths. The audio on the path from
google talk to FS results in about 3-5 seconds delay.

Does anybody experience the similar issue or can help me to point out where
could be the problem?

Thanks.

Chaofeng
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From brian at freeswitch.org  Mon Mar  8 09:21:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 11:21:48 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
Message-ID: <8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>

ok you aren't catching one key thing here.. you no longer need two profiles.

/b

On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:

> Hi again,
> 
> Thanks for the help so far.  Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
> 
> I updated to build 16938 by means of "make current"
> 
> I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
> 
> I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on.  I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
> 
> I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
> 
> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 
> One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
> 
> Any further ideas where I am going wrong here please?
> 
> thanks again in advance
> Bruce



From msc at freeswitch.org  Mon Mar  8 09:22:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 09:22:01 -0800
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
Message-ID: <87f2f3b91003080922o34d5495foc9641bfc3357989@mail.gmail.com>

On Mon, Mar 8, 2010 at 8:55 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> application,eavesdrop,mod_dptools,/opt/freeswitch/mod/mod_dptools.so
>
> It is an app, not an API.
>

I other words use session:execute("eavesdrop") and not pbx:api("eavesdrop")
-MC
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From t.mahe at telemaque.fr  Mon Mar  8 09:23:55 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 08 Mar 2010 18:23:55 +0100
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
Message-ID: <4B9532AB.5030407@telemaque.fr>

+1 with Brian,

Just as I have 5 spare minutes and to be kind:


* E1 controller config:

controller E1 0/0/0
 framing NO-CRC4
 pri-group timeslots 1-31
 description E1_1
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn not-end-to-end 64
 isdn incoming-voice voice
 isdn bchan-number-order descending round-robin
 no cdp enable
!
voice-port 0/0/0:15
 no comfort-noise
 cptone FR
 timeouts call-disconnect 1
 timeouts wait-release 1
!

* Dial-peers ( SIP to E1 and E1 to SIP):

dial-peer voice 100 pots
 description SIP_TO_E1
 destination-pattern 0........
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 progress_ind disconnect enable 8
 no digit-strip
 port 0/0/0:15
!
dial-peer voice 500 voip
 description E1_TO_SIP
 destination-pattern ....
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:FREESWITCH_IP
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!


Remember to adapt this config to your needs ( there are mandatory
missing parts, but at least you have some config options to look for).

Regards,

Gled.

Brian West a ?crit :
> You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.
> 
> /b
> 
> On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> 
>> HI,
>>  
>> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
>>  
>> Thanks
>> Srinivas
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From paul.gore.j at gmail.com  Mon Mar  8 09:22:37 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 8 Mar 2010 12:22:37 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
Message-ID: <da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>

It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will
only need to send one extension back..."
In my real dial plan I do use unique names for the extensions, but as I
mentioned FS only executes first extension from the response. If I place
same extensions in a static XML they work as  expected.
I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you
post a sample response you generate, please?


On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com>wrote:

>
> Hi
>
> "As per FS wiki we can only return one XML extension upon FS request. "
>
> Could you please tell me where did you find that info? I do return several
> extensions by request and FS works as expect (Am I wrong doing that?)
>
> "Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?"
>
> Yes, It seems to be possible (Am I wrong?) but notice that you are giving
> the  same extension name for all your extensions:
>
>  <extension name="test9">
>
>
> I think this can mislead dialplan_hunt in mod_dialplan_xml...
> Give each extension a unique name, and FS will parse/regex all them for
> you...
>
>
> All the best,
> Paulo
>
> > From: brian at freeswitch.org
> > Date: Sun, 7 Mar 2010 20:17:19 -0600
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] XML curl and multiple extension in
> dialplan
>
> >
> > Sounds like you zigged where you should have zagged.
> >
> > /b
> >
> > On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> >
> > > :-))
> > > I used it a lot. Unfortunately we do not store did number as a field,
> as I said we store complete extension XML, with extension number and context
> as a key.
> > > So essentially I will retrieve a bunch of extension XMLs which I need
> to parse and run against regex matcher to find out the one, which is what
> exactly FS is doing when XML dial plan is defined in a static file.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 09:32:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 11:32:05 -0600
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <4B9532AB.5030407@telemaque.fr>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr>
Message-ID: <23FE228D-9CA2-4858-8A94-5AD5FD770FA8@freeswitch.org>

Thank you for taking the time to help out... maybe a wiki page can be created on this moving forward.  I wasn't in any way trying to be mean or disrespectful.  I'm clearly trying to state that we can't possibly help everyone with everything at some point it just starts to get overwhelming to try to help everyone.  I thank everyone that are starting to step up and help people out on the list.  Lets keep that up it helps greatly.

Thanks,
Brian

On Mar 8, 2010, at 11:23 AM, Tristan Mah? wrote:

> +1 with Brian,
> 
> Just as I have 5 spare minutes and to be kind:



From will.traenkle at yahoo.com  Mon Mar  8 10:05:24 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 10:05:24 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <756904.84769.qm@web57613.mail.re1.yahoo.com>

I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

My hunt group code is below:


session.answer();
var domain_name = session.getVariable("domain_name");
var extension = '7000';
var result;
var timeoutpin = 7500;

var objdate = new Date();
var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours
var adjustoperator = "-"; //+ or -
if (adjustoperator == "-") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
if (adjustoperator == "+") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
var Hours = objdate2.getHours();
var Mins = objdate2.getMinutes();
var Seconds = objdate2.getSeconds();
var Month = objdate2.getMonth() + 1;
var Date = objdate2.getDate();
var Year = objdate2.getYear()
var Day = objdate2.getDay()+1;
var exit = false;

 function get_sofia_contact(extension,domain_name, profile){
        if (profile == "auto") {
                        profile = "internal";
                        session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                        sofia_contact = session.getVariable("sofia_contact_"+extension);

                        if (sofia_contact == "error/user_not_registered") {
                                profile = "external";
                                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                                sofia_contact = session.getVariable("sofia_contact_"+extension);
                        }
        }
        else {
                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                sofia_contact = session.getVariable("sofia_contact_"+extension);
        }
        console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" );
        return sofia_contact;
 }


 function mycb( session, type, obj, arg ) {
    try {
        if ( type == "dtmf" ) {
          console_log( "info", "digit: "+obj.digit+"\n" );
          if ( obj.digit == "#" ) {
            //console_log( "info", "detected pound sign.\n" );
            exit = true;
            return( false );
          }

          dtmf.digits += obj.digit;

          if ( dtmf.digits.length >= digitmaxlength ) {
            exit = true;
            return( false );
          }
        }
    } catch (e) {
        console_log( "err", e+"\n" );
    }
    return( true );
 } //end function mycb


dialed_extension = session.getVariable("dialed_extension");
domain_name = session.getVariable("domain_name");
domain = session.getVariable("domain");
us_ring = session.getVariable("us-ring");
caller_id_name = session.getVariable("caller_id_name");
caller_id_number = session.getVariable("caller_id_number");
effective_caller_id_name = session.getVariable("effective_caller_id_name");
effective_caller_id_number = session.getVariable("effective_caller_id_number");
outbound_caller_id_name = session.getVariable("outbound_caller_id_name");
outbound_caller_id_number = session.getVariable("outbound_caller_id_number");
session.execute("set", "ringback=${hold_music}");          //set to ringtone
session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone
session.execute("set", "call_timeout=30");
session.execute("export", "call_timeout=30");
session.execute("set", "continue_on_fail=true");
session.execute("set", "hangup_after_bridge=true");

//console_log( "info", "dialed extension:"+dialed_extension+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "us_ring: "+us_ring+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "domain_name: "+domain_name+".\n" );

//console_log( "info", "action call now don't wait for dtmf\n" );
if ( session.ready() ) {
        session.answer();

        sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto");

        session.execute("bridge", sofia_contact_1000);
        session.execute("bridge", "sofia/gateway/flowroute.com/16194548924");

        //timeout
        if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){
                session.execute("info", "");
                session.execute("transfer", "*991000");
        }

        //clear variables
        dialed_extension = "";
        new_extension = "";
        domain_name = "";
        domain = "";
} //end if session.ready


Thanks.

-Will


      
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From lloyd.aloysius at gmail.com  Mon Mar  8 10:23:54 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 13:23:54 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer Help
Message-ID: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>

Hi All,

I would like to setup a dial plan, Dial *99[extension number]  send a call
directly to voicemail.

 I setup the following Dial Plan . But when I transfer the call
*99[extension number] the call failed.

    <extension name="voicemail_transfer">
      <condition field="destination_number" expression="^(\*9920[0-9])$">
  <action application="answer"/>
  <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
  </condition>
    </extension>
But when I transfer the call *99[extension number] the call failed.My
extensions range 201 - 209

Please let me know where the mistake is.

Thanks
Lloyd
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From vizentini at hotmail.com  Mon Mar  8 10:31:39 2010
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Mon, 8 Mar 2010 18:31:39 +0000
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>,
	<4B9409FF.5060608@gmail.com>,
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>,
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>,
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>,
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>,
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>,
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>,
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>,
	<da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
Message-ID: <SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>


Hello,
FreeSWITCH version: 1.0.trunk (16870M)
I have replied FS even with extensions you have emailed here (in my case I have data="sofia/gateway/myGW/888 at conference.freeswitch.org/888") - it works
You may need to give a look at "CS_ROUTING" state.

best,PauloDate: Mon, 8 Mar 2010 12:22:37 -0500
From: paul.gore.j at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan

It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will only need to send one extension back..."
In my real dial plan I do use unique names for the extensions, but as I mentioned FS only executes first extension from the response. If I place same extensions in a static XML they work as  expected.
I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you post a sample response you generate, please?


On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com> wrote:







Hi
"As per FS wiki we can only return one XML extension upon FS request. "
Could you please tell me where did you find that info? I do return several extensions by request and FS works as expect (Am I wrong doing that?)

"Now we need to return a set of extensions with same number
condition and context but with different second condition so that FS
can pick up the right one. 
Is it possible at all? And if yes how response XML with multiple extensions should look like?"
Yes, It seems to be possible (Am I wrong?) but notice that you are giving the  same extension name for all your extensions:

 <extension name="test9">

I think this can mislead dialplan_hunt in mod_dialplan_xml...Give each extension a unique name, and FS will parse/regex all them for you... 

All the best,
Paulo
> From: brian at freeswitch.org
> Date: Sun, 7 Mar 2010 20:17:19 -0600
> To: freeswitch-users at lists.freeswitch.org

> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
> 
> Sounds like you zigged where you should have zagged.
> 
> /b
> 
> On Mar 7, 2010, at 8:03 PM, paul gore wrote:

> 
> > :-))
> >  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> > So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.

> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org

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

> http://www.freeswitch.org
 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now.


_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org



 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From gavin.henry at gmail.com  Mon Mar  8 10:33:27 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 18:33:27 +0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
	<13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
	<70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>
Message-ID: <13ca621c1003081033n5ab07ea6lf3233194d4027fb3@mail.gmail.com>

 On 8 March 2010 11:30, Jay Binks <jaybinks at gmail.com> wrote:
> The qn here is do you bypass media or do you do complete topology
> hiding ??
>
> Sip only is kind of a given , ?it gets harder if the virtualized pbx
> does media.

The pbx does the media here and then on to our VoIP provider platform.


From brian at freeswitch.org  Mon Mar  8 10:33:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 12:33:26 -0600
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <0419C3ED-7F68-4299-8170-39C376FE3CC0@freeswitch.org>

That looks correct can you show me console output?

/b

On Mar 8, 2010, at 12:23 PM, Aloysius Lloyd wrote:

> Hi All,
> 
> I would like to setup a dial plan, Dial *99[extension number]  send a call directly to voicemail.
> 
>  I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed.
> 
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
> 	  <action application="answer"/>
> 	  <action application="sleep" data="1000"/>
> 	  <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
> 	  </condition>
>     </extension>
> 	
> But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209
> 
> Please let me know where the mistake is.
> 
> Thanks
> Lloyd

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From chris.chen2004 at gmail.com  Mon Mar  8 10:34:57 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 13:34:57 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <507898381003081034t13752159y87c2deb138fbb5f6@mail.gmail.com>

Lloyd, you didn't define ${dialed_extension}, before you call the voicemail
application in your dialplan:
 <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>

Please fix that.

Chris

On Mon, Mar 8, 2010 at 1:23 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
>
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
>   <action application="answer"/>
>   <action application="sleep" data="1000"/>
>   <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>   </condition>
>     </extension>
> But when I transfer the call *99[extension number] the call failed.My
> extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Mon Mar  8 10:34:15 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 8 Mar 2010 12:34:15 -0600
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <95DBAAA1-EB57-4440-8C7B-618053730DB5@gmail.com>

Much easier to see what's going on with logs.  Post the snippet of a  
test call from your freeswitch logs in debug to http://pastebin.freeswitch.org/ 
  and share the link.


On Mar 8, 2010, at 12:23 PM, Aloysius Lloyd wrote:

> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send  
> a call directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call  
> *99[extension number] the call failed.
>
>     <extension name="voicemail_transfer">
>       <condition field="destination_number"  
> expression="^(\*9920[0-9])$">
> 	  <action application="answer"/>
> 	  <action application="sleep" data="1000"/>
> 	  <action application="voicemail" data="default ${domain_name} $ 
> {dialed_extension}"/>
> 	  </condition>
>     </extension>
> 	
> But when I transfer the call *99[extension number] the call  
> failed.My extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at carmickle.com  Mon Mar  8 10:40:25 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 8 Mar 2010 13:40:25 -0500
Subject: [Freeswitch-users] hunt group - play music when trying	external
	numbers
In-Reply-To: <756904.84769.qm@web57613.mail.re1.yahoo.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
Message-ID: <20100308184025.GL18427@base.carmickle.com>

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC


From frank at carmickle.com  Mon Mar  8 10:45:17 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 8 Mar 2010 13:45:17 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail
	Transfer	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <20100308184517.GM18427@base.carmickle.com>

Hello

On Mon, Mar 08, Aloysius Lloyd wrote:
> Hi All,
> 
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
> 
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
> 
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">

Should be

<condition field="destination_number" expression="^\*99(20[0-9])$">

>   <action application="answer"/>
>   <action application="sleep" data="1000"/>
>   <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>

and

<action application="voicemail" data="default ${domain_name} $1"/>

>   </condition>
>     </extension>

HTH
--FC


From dave at 3c.co.uk  Mon Mar  8 10:46:22 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 15:46:22 -0300
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <AF01ECDC33A74538AEB69214232F3E51@DELL9>

Hi Aloysius,

Best guess is that
<action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
isn't what you meant - ${dialed_extension} will be 9920x, not 20x.

You can fix this by changing the regex to
<condition field="destination_number" expression="^\*99(20[0-9])$">
(the bit in ()s will end up in $1) and the line above to
<action application="voicemail" data="default ${domain_name} $1"/>

Cheers --

Dave
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 3:23 PM
  Subject: [Freeswitch-users] Send a call directly to voicemail Transfer Help


  Hi All, 


  I would like to setup a dial plan, Dial *99[extension number]  send a call directly to voicemail.


   I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed.


      <extension name="voicemail_transfer">
        <condition field="destination_number" expression="^(\*9920[0-9])$">
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
   </condition>
      </extension>
  But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209


  Please let me know where the mistake is.


  Thanks
  Lloyd


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


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From grevenx at me.com  Mon Mar  8 10:52:57 2010
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Mon, 08 Mar 2010 19:52:57 +0100
Subject: [Freeswitch-users] FreeSWITCH cluster with IM/Presence
Message-ID: <9FD45932-DD45-4370-A08B-9C06C6D3289C@me.com>

Hi.

I currently have a project where I'm researching how to establish a clustered and scalable platform with:
- registration
- presence/blf between eyebeam/bria and Linksys SPA962 clients
- IM between eyebeam/bria clients.
- calling between the clients

The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". 
How the registrations and calls etc. are balanced between them is not decided
yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here?
The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers
(which can be scaled by adding n-nodes).

I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, 
and what it would take to make something like this work. If there's any part of FS that is lacking some implementation
to make such a scenario work, we are interested in providing the funding to support those features.
I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would
be helpful in this scenario or not.

This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers 
to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to 
be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to.

Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence.
I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform.


Best regards,
Even Andr? Fiskvik
CTO - Oyatel AS



From grevenx at me.com  Mon Mar  8 11:11:42 2010
From: grevenx at me.com (=?windows-1252?Q?Even_Andr=E9_Fiskvik?=)
Date: Mon, 08 Mar 2010 20:11:42 +0100
Subject: [Freeswitch-users] server to server jingle
In-Reply-To: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
References: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
Message-ID: <147756F3-FD28-45C3-BF2E-DF1DA6399CAC@me.com>

Perhaps you should write a bit about what you're exactly trying to achieve and
if it doesn't seem to "work as expected", describe what issues you are having?

Best regards,
Even Andr? Fiskvik

On 2. mars 2010, at 05.38, Madovsky wrote:

> Hi all,
>  
> is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
> I read this article
> http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
> but no example
> and this one
> http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
> but seems to doesn't work as expected.
>  
> Thanks !
>  
> Franck
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Mon Mar  8 11:26:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 11:26:06 -0800
Subject: [Freeswitch-users] Conference Recording
In-Reply-To: <4B948B0E.8050909@todandlorna.com>
References: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
	<4B948B0E.8050909@todandlorna.com>
Message-ID: <87f2f3b91003081126q591662eq476604ddc5c51afd@mail.gmail.com>

Sorry, I still need to x-code and upload Tod's wonderful discussion of
mod_fifo. :)
-MC

On Sun, Mar 7, 2010 at 9:28 PM, Tod Hansmann <freeswitch.org at todandlorna.com
> wrote:

>  For the whole thing?  I know parts were recorded by different people.  I'm
> not sure about the whole thing.
>
> -Tod Hansmann
>
>
> On 3/7/2010 5:28 PM, Josh Rivers wrote:
>
> On last week's conference call, someone was recording, did that recording
> get put online somewhere?
>
>  Josh
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Mar  8 11:27:14 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 14:27:14 -0500
Subject: [Freeswitch-users] server to server jingle
References: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
	<147756F3-FD28-45C3-BF2E-DF1DA6399CAC@me.com>
Message-ID: <2161970E983E4410A1695D4D4E64430E@MOBILEE1705>

I wanted only to avoid to write an xml for every client who uses xmpp.
nevermind, I'm too novice on Openfire and don't want to use anymore time to learn it
since I need to learn more FS.

Thanks for your answer

Franck
  ----- Original Message ----- 
  From: Even Andr? Fiskvik 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 2:11 PM
  Subject: Re: [Freeswitch-users] server to server jingle


  Perhaps you should write a bit about what you're exactly trying to achieve and
  if it doesn't seem to "work as expected", describe what issues you are having?


  Best regards,
  Even Andr? Fiskvik


  On 2. mars 2010, at 05.38, Madovsky wrote:


    Hi all,

    is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
    I read this article
    http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
    but no example
    and this one
    http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
    but seems to doesn't work as expected.

    Thanks !

    Franck

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





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


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From lloyd.aloysius at gmail.com  Mon Mar  8 11:28:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 14:28:37 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <AF01ECDC33A74538AEB69214232F3E51@DELL9>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
	<AF01ECDC33A74538AEB69214232F3E51@DELL9>
Message-ID: <8a19bf2e1003081128v55a993a4m42d6860bf6657162@mail.gmail.com>

Thank you for all the suggestions.

David & Frank suggestion solve the problem. Here is the working dial plan

  <extension name="voicemail_transfer">
<condition field="destination_number" expression="^\*99(20[0-9])$">
          <action application="answer"/>
          <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain_name} $1"/>
         </condition>
         </extension>


Thank you again.

Regards.
Lloyd


On Mon, Mar 8, 2010 at 1:46 PM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Aloysius,
>
> Best guess is that
> <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
> isn't what you meant - ${dialed_extension} will be 9920x, not 20x.
>
> You can fix this by changing the regex to
> <condition field="destination_number" expression="^\*99(20[0-9])$">
> (the bit in ()s will end up in $1) and the line above to
> <action application="voicemail" data="default ${domain_name} $1"/>
>
> Cheers --
>
> Dave
>
> ----- Original Message -----
> *From:* Aloysius Lloyd <lloyd.aloysius at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 08, 2010 3:23 PM
> *Subject:* [Freeswitch-users] Send a call directly to voicemail Transfer
> Help
>
> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
>
>      <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
>  <action application="answer"/>
>  <action application="sleep" data="1000"/>
>  <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>  </condition>
>     </extension>
>  But when I transfer the call *99[extension number] the call failed.My
> extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Mon Mar  8 11:42:12 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 19:42:12 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion on
	Polycom IP6000 handsets
Message-ID: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>

This is almost certainly more of a Polycom issue than a FreeSWITCH one
but I'll mention it here on the offchance that someone else has
experienced the following:

Loud G.711a calls that are being transcoded to G.722 are massively
distorted on Polycom IP6000 handsets and on all other handsets the
call just sounds loud and isn't distorted. This is most common in
conference calls where callers are using a mixture of codecs.

If I disable G.722 the Polycom sounds fine even though the G.711a call
is still too loud and if I lower the volume of the conference the
distortion still happens.

So, has anyone else experienced this and if so, did you find a solution?

Regards,
Dan Lane


From m.sobkow at marketelsystems.com  Mon Mar  8 11:46:09 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Mar 2010 13:46:09 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
Message-ID: <4B955401.1030004@marketelsystems.com>

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From brian at freeswitch.org  Mon Mar  8 11:51:39 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 13:51:39 -0600
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
Message-ID: <90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>

I'm going to guess you're using 1.0.4?

/b

On Mar 8, 2010, at 1:42 PM, Dan Lane wrote:

> This is almost certainly more of a Polycom issue than a FreeSWITCH one
> but I'll mention it here on the offchance that someone else has
> experienced the following:
> 
> Loud G.711a calls that are being transcoded to G.722 are massively
> distorted on Polycom IP6000 handsets and on all other handsets the
> call just sounds loud and isn't distorted. This is most common in
> conference calls where callers are using a mixture of codecs.
> 
> If I disable G.722 the Polycom sounds fine even though the G.711a call
> is still too loud and if I lower the volume of the conference the
> distortion still happens.
> 
> So, has anyone else experienced this and if so, did you find a solution?
> 
> Regards,
> Dan Lane
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Mon Mar  8 12:04:26 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 8 Mar 2010 15:04:26 -0500
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
Message-ID: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>

You could use the api:

 FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
string apiResult = string.Empty;
string Uuid;
string NumberToDial = "3475558308";
string OutgoingCallerID = "2155556666";

Uuid = fsApi.ExecuteString("create_uuid");

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
Uuid, OutgoingCallerID, NumberToDial));






On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:

> Hi,
> I am trying to create a mod_managed API application that takes 2 phone
> numbers as params.  These numbers should then be used to make 2 calls and to
> bridge them. The issue i am having is working out how to place the first
> call leg from in mod_managed.  All the methods i have found are for the
> second leg and require you to pass in the first call leg.
>
> I know there are easier way to do this out of the managed code but i want
> it in the managed code as this is just a proof of concept to prove we can do
> a callback like system via managed code to be hooked into other external
> systems.
> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>
> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
> while (s.ready()) {
> // The call has been answered
> }
>
> This seems to be exactly what i am looking for but seems to be missing from
> mod_managed. I am hoping someone can tell me how to do something similar in
> mod_managed since even if it is not part of the managed code mod_managed is
> meant to have the native access as well so i would assume it would be
> possible.
>
> Thanks in advance
>
> Chris
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From matt at packetfoundry.net  Mon Mar  8 09:27:42 2010
From: matt at packetfoundry.net (Matt Putnam)
Date: Mon, 08 Mar 2010 11:27:42 -0600
Subject: [Freeswitch-users] mod_spidermonkey
Message-ID: <4B95338E.8030709@packetfoundry.net>

http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel

Has anyone found a solution to the above issue? I am having the same 
issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.


Thanks,
Matt


From brian at freeswitch.org  Mon Mar  8 12:33:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 14:33:52 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B95338E.8030709@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
Message-ID: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>

cd /usr/local/freeswitch/mod
ldd mod_spidermonkey.so

And post it to that jira.

/b

On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:

> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
> 
> Has anyone found a solution to the above issue? I am having the same 
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
> 
> 
> Thanks,
> Matt

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From chris.chen2004 at gmail.com  Mon Mar  8 12:39:19 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 15:39:19 -0500
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B95338E.8030709@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
Message-ID: <507898381003081239m4196f0ccn81dc7bbc2f8874f6@mail.gmail.com>

I am having the same problem as reported in the JIRA, although this has been
about a week. Still the same on latest SVN trunk.
I am running Centos 5.4 X86_64.
Thanks,

Chris

On Mon, Mar 8, 2010 at 12:27 PM, Matt Putnam <matt at packetfoundry.net> wrote:

>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chris.chen2004 at gmail.com  Mon Mar  8 12:43:19 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 15:43:19 -0500
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
Message-ID: <507898381003081243j2d73456ei4b04fda5c8fbfef3@mail.gmail.com>

Hi Brian, I added the information from my Centos 5.4 to the JIRA, please see
the comment.
Thanks,
Chris

On Mon, Mar 8, 2010 at 3:33 PM, Brian West <brian at freeswitch.org> wrote:

> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
> And post it to that jira.
>
> /b
>
> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Mon Mar  8 12:46:30 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 20:46:30 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
Message-ID: <b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>

On Mon, Mar 8, 2010 at 7:51 PM, Brian West <brian at freeswitch.org> wrote:
> I'm going to guess you're using 1.0.4?

I am, was this fixed in 1.0.5?

While I have 1.0.5 boxes (which identify themselves as 1.0.trunk)
coming out of my ears the Polycoms are all overseas and in-use so
getting them temporarily connected to a 1.0.5 box is troublesome.


From paul.gore.j at gmail.com  Mon Mar  8 12:48:56 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 8 Mar 2010 15:48:56 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
	<da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
	<SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>
Message-ID: <da4ab26e1003081248l37fa5716t72d36a897747a5a5@mail.gmail.com>

Thanks a lot Paulo!
It does work, we have updated to the latest SVN and checked cs_routing and
apparently discovered a problem with response generating script, it wasn't
updated after recent DB change.


On Mon, Mar 8, 2010 at 1:31 PM, Paulo Vicentini <vizentini at hotmail.com>wrote:

>  Hello,
>
> FreeSWITCH version: 1.0.trunk (16870M)
>
> I have replied FS even with extensions you have emailed here (in my case I
> have data="sofia/gateway/myGW/888 at conference.freeswitch.org/888") - it
> works
>
> You may need to give a look at "CS_ROUTING" state.
>
>
> best,
> Paulo
> ------------------------------
> Date: Mon, 8 Mar 2010 12:22:37 -0500
> From: paul.gore.j at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
>
> It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will
> only need to send one extension back..."
> In my real dial plan I do use unique names for the extensions, but as I
> mentioned FS only executes first extension from the response. If I place
> same extensions in a static XML they work as  expected.
> I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you
> post a sample response you generate, please?
>
>
> On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com>wrote:
>
>
> Hi
>
> "As per FS wiki we can only return one XML extension upon FS request. "
>
> Could you please tell me where did you find that info? I do return several
> extensions by request and FS works as expect (Am I wrong doing that?)
>
> "Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?"
>
> Yes, It seems to be possible (Am I wrong?) but notice that you are giving
> the  same extension name for all your extensions:
>
>  <extension name="test9">
>
>
> I think this can mislead dialplan_hunt in mod_dialplan_xml...
> Give each extension a unique name, and FS will parse/regex all them for
> you...
>
>
> All the best,
> Paulo
>
> > From: brian at freeswitch.org
> > Date: Sun, 7 Mar 2010 20:17:19 -0600
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] XML curl and multiple extension in
> dialplan
>
> >
> > Sounds like you zigged where you should have zagged.
> >
> > /b
> >
> > On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> >
> > > :-))
> > > I used it a lot. Unfortunately we do not store did number as a field,
> as I said we store complete extension XML, with extension number and context
> as a key.
> > > So essentially I will retrieve a bunch of extension XMLs which I need
> to parse and run against regex matcher to find out the one, which is what
> exactly FS is doing when XML dial plan is defined in a static file.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From matt at packetfoundry.net  Mon Mar  8 12:49:38 2010
From: matt at packetfoundry.net (Matt Putnam)
Date: Mon, 08 Mar 2010 14:49:38 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
Message-ID: <4B9562E2.7000207@packetfoundry.net>

Posted mine up as well.

On 3/8/2010 2:33 PM, Brian West wrote:
> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
> And post it to that jira.
>
> /b
>
> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>>
>> Has anyone found a solution to the above issue? I am having the same
>> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>>
>>
>> Thanks,
>> Matt
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From brian at freeswitch.org  Mon Mar  8 13:19:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:19:18 -0600
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
	<b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
Message-ID: <C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>

It was a bug in spandsp codecs thats been fixed for sometime.

/b

On Mar 8, 2010, at 2:46 PM, Dan Lane wrote:

> I am, was this fixed in 1.0.5?
> 
> While I have 1.0.5 boxes (which identify themselves as 1.0.trunk)
> coming out of my ears the Polycoms are all overseas and in-use so
> getting them temporarily connected to a 1.0.5 box is troublesome.



From javieraristizabal at gmail.com  Mon Mar  8 13:38:39 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:38:39 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
Message-ID: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>

Hi folks!!
I have an incoming call being bridged to an outgoing call ann i'm trying to
pass the phone number of the incoming caller as the caller id on the leg B.
But it doesn't work.

Here is my dialplan:

<extension name="incoming">
        <condition field="destination_number" expression="^(3055908934)$">
                <action application="info"/>
                <action application="set"
data="effective_callee_id_number=${caller_id_number}"/>
                <action application="bridge"
data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
         </condition>
    </extension>

Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368


This is the INVITE on the bLeg:

INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
Max-Forwards: 68
From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
;transport=udp>;tag=K6U13Hy2rpgBe
To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
>
Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
CSeq: 127923447 INVITE
Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 250
X-FS-Support: update_display
Remote-Party-ID: "Javier"
<sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
s=FreeSWITCH
c=IN IP4 66.231.242.173
t=0 0
m=audio 23516 RTP/AVP 18 101 13
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20


And i'm on FreeSWITCH Version 1.0.trunk (16921)


Many thanks


/Javier
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From null at invalid.name  Mon Mar  8 13:41:53 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 21:41:53 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
	<b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
	<C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>
Message-ID: <b042b23b1003081341o78b23247jad74aabf00c82a76@mail.gmail.com>

On Mon, Mar 8, 2010 at 9:19 PM, Brian West <brian at freeswitch.org> wrote:
> It was a bug in spandsp codecs thats been fixed for sometime.

Thanks for the quick feedback... I'll try and get that box upgraded ASAP :)


From dule.maillist at gmail.com  Mon Mar  8 13:50:26 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 8 Mar 2010 16:50:26 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
Message-ID: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>

Try using the 'export' app rather than 'set'.

Dan

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Hi folks!!
> I have an incoming call being bridged to an outgoing call ann i'm trying to
> pass the phone number of the incoming caller as the caller id on the leg B.
> But it doesn't work.
>
> Here is my dialplan:
>
> <extension name="incoming">
>         <condition field="destination_number" expression="^(3055908934)$">
>                 <action application="info"/>
>                 <action application="set"
> data="effective_callee_id_number=${caller_id_number}"/>
>                 <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
>          </condition>
>     </extension>
>
> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
>
>
> This is the INVITE on the bLeg:
>
> INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
>  Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
> Max-Forwards: 68
> From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
> ;transport=udp>;tag=K6U13Hy2rpgBe
> To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
> >
> Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
> CSeq: 127923447 INVITE
> Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 250
> X-FS-Support: update_display
> Remote-Party-ID: "Javier" <sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
> s=FreeSWITCH
> c=IN IP4 66.231.242.173
> t=0 0
> m=audio 23516 RTP/AVP 18 101 13
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
>
> And i'm on FreeSWITCH Version 1.0.trunk (16921)
>
>
> Many thanks
>
>
> /Javier
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 13:52:02 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:52:02 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
Message-ID: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>

chances are your provider wants the caller id in the from field.

/b

On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:

> Hi folks!! 
> I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
> 



From Mailings at kh-dev.de  Mon Mar  8 13:54:10 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 22:54:10 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>

Hi

you could also use newest trunk of t38modem + trunk of OPAL.
The newest trunk version(s) use SpanDSP as fax engine and it works more reliable than the previous t38 engine (I think it was integrated in OPAL).
It also supports fallback to t30 audio fax if you're connected to ISDN network.

Together with HylaFAX+ the success rate is ok (unfortunately not perfect).

Klaus

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Monday, March 08, 2010 5:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

On Mon, Mar 8, 2010 at 10:57 AM, Steve Underwood <steveu at coppice.org> wrote:
> We already have a better FAX engine than that. What we struggle to find
> is the time to properly integrate it with FS.
>
> Steve

  Thank you, that's what I was expecting.

  Without going into grueling detail it isn't clear why T.38 gateway
functionality (for example) is missing in FreeSWITCH.  I know SpanDSP
is available and most likely has this ability.  I just wasn't sure why
it's missing in FS.  This is also a mere curiosity for me; I don't
have much interest in fax functionality and I don't follow it too
closely.

  Thanks again for the response and explanation.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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




From javieraristizabal at gmail.com  Mon Mar  8 13:58:46 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:58:46 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
Message-ID: <b6f11f301003081358t4ba2d935l9ce408a7425fa5d1@mail.gmail.com>

HI Dan. Thanks.  I did it with the same result.


/Javier

On Mon, Mar 8, 2010 at 4:50 PM, Dan Le <dule.maillist at gmail.com> wrote:

> Try using the 'export' app rather than 'set'.
>
> Dan
>
> 2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Hi folks!!
>> I have an incoming call being bridged to an outgoing call ann i'm trying
>> to pass the phone number of the incoming caller as the caller id on the leg
>> B. But it doesn't work.
>>
>> Here is my dialplan:
>>
>> <extension name="incoming">
>>         <condition field="destination_number" expression="^(3055908934)$">
>>                 <action application="info"/>
>>                 <action application="set"
>> data="effective_callee_id_number=${caller_id_number}"/>
>>                 <action application="bridge"
>> data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
>>          </condition>
>>     </extension>
>>
>> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
>>
>>
>> This is the INVITE on the bLeg:
>>
>> INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
>>  Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
>> Max-Forwards: 68
>> From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
>> ;transport=udp>;tag=K6U13Hy2rpgBe
>> To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
>> >
>> Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
>> CSeq: 127923447 INVITE
>> Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 250
>> X-FS-Support: update_display
>> Remote-Party-ID: "Javier" <sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
>> >;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
>> s=FreeSWITCH
>> c=IN IP4 66.231.242.173
>> t=0 0
>> m=audio 23516 RTP/AVP 18 101 13
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>>
>> And i'm on FreeSWITCH Version 1.0.trunk (16921)
>>
>>
>> Many thanks
>>
>>
>> /Javier
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 13:59:30 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:59:30 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
Message-ID: <BDACCFA5-0901-4B62-80BE-3EAF5FAAEC05@freeswitch.org>

No we by default already pass it... no need to export vs set.. plus effective_caller_id_number/name is set on the A-LEG and auto exported for you to the B-LEG when created.

His idiot provider wants it in the from field I suspect.

/b

On Mar 8, 2010, at 3:50 PM, Dan Le wrote:

> Try using the 'export' app rather than 'set'.
> 
> Dan



From javieraristizabal at gmail.com  Mon Mar  8 13:59:47 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:59:47 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
Message-ID: <b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>

Hi Brian. Thanks, yeah it is possible. Is there a way to do that?

/Javier

On Mon, Mar 8, 2010 at 4:52 PM, Brian West <brian at freeswitch.org> wrote:

> chances are your provider wants the caller id in the from field.
>
> /b
>
> On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:
>
> > Hi folks!!
> > I have an incoming call being bridged to an outgoing call ann i'm trying
> to pass the phone number of the incoming caller as the caller id on the leg
> B. But it doesn't work.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Mon Mar  8 13:59:34 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 08:59:34 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com> 
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
Message-ID: <9394d42f1003081359u5cb0f2e0q90695d1abb0be9b4@mail.gmail.com>

Thanks phillip.
I assume i could then use the command new ManagedSession(uuid); to get
access to the session after it is created.  I will give it a go later today
but i think you are right that should work.

For a mater of interest would you have any idea where in the source code for
FS i could find the implementation for the originate method called via the
API command you just suggested?  If the implementation is not too
complicated I may look at adding it to the managed code as the managed code
is meant to have access to the native APIs so it should be able to issue the
same commands i would have thought. If it is impossible i will stick with
the way you just suggested.

Thanks

Chris

On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> You could use the api:
>
>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
> string apiResult = string.Empty;
> string Uuid;
> string NumberToDial = "3475558308";
> string OutgoingCallerID = "2155556666";
>
> Uuid = fsApi.ExecuteString("create_uuid");
>
> apiResult = fsApi.Execute("originate",
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
> Uuid, OutgoingCallerID, NumberToDial));
>
>
>
>
>
>
> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Hi,
>> I am trying to create a mod_managed API application that takes 2 phone
>> numbers as params.  These numbers should then be used to make 2 calls and to
>> bridge them. The issue i am having is working out how to place the first
>> call leg from in mod_managed.  All the methods i have found are for the
>> second leg and require you to pass in the first call leg.
>>
>> I know there are easier way to do this out of the managed code but i want
>> it in the managed code as this is just a proof of concept to prove we can do
>> a callback like system via managed code to be hooked into other external
>> systems.
>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>
>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>> while (s.ready()) {
>> // The call has been answered
>> }
>>
>> This seems to be exactly what i am looking for but seems to be missing
>> from mod_managed. I am hoping someone can tell me how to do something
>> similar in mod_managed since even if it is not part of the managed code
>> mod_managed is meant to have the native access as well so i would assume it
>> would be possible.
>>
>> Thanks in advance
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar  8 14:00:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:00:13 -0800
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
Message-ID: <87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>

On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:

> chances are your provider wants the caller id in the from field.
>
> /b
>
This sounds more likely. Javar tried the export already and that didn't
help. What is the proper procedure for setting the From field? Can you just
export sip_from_user when doing the bridge?
-MC


>
> On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:
>
> > Hi folks!!
> > I have an incoming call being bridged to an outgoing call ann i'm trying
> to pass the phone number of the incoming caller as the caller id on the leg
> B. But it doesn't work.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  8 14:02:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:02:26 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
Message-ID: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>

Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).



From brian at freeswitch.org  Mon Mar  8 14:03:49 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:03:49 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>
Message-ID: <029A5E59-4C83-451F-91AB-465C4585EBFA@freeswitch.org>

Try sip_from_uri

/b

On Mar 8, 2010, at 4:00 PM, Michael Collins wrote:

> 
> 
> On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
> chances are your provider wants the caller id in the from field.
> 
> /b
> This sounds more likely. Javar tried the export already and that didn't help. What is the proper procedure for setting the From field? Can you just export sip_from_user when doing the bridge?
> -MC

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From brian at freeswitch.org  Mon Mar  8 14:04:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:04:33 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
Message-ID: <EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>

if all else fails try reading do_invite in sofia_glue.c you can really change just about ANYTHING to do with the invite with variables... be careful some let you totally break the packet where it won't even work.  :P

/b

On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:

> Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
> 
> /Javier



From javieraristizabal at gmail.com  Mon Mar  8 14:12:39 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:12:39 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
Message-ID: <b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>

Thanks Brian, let me try your ideas.

/Javier

On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian at freeswitch.org> wrote:

> if all else fails try reading do_invite in sofia_glue.c you can really
> change just about ANYTHING to do with the invite with variables... be
> careful some let you totally break the packet where it won't even work.  :P
>
> /b
>
> On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:
>
> > Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
> >
> > /Javier
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From will.traenkle at yahoo.com  Mon Mar  8 14:23:45 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 14:23:45 -0800 (PST)
Subject: [Freeswitch-users] auto attendant -> hunt group -> external number
	does not ring
Message-ID: <302537.41702.qm@web57609.mail.re1.yahoo.com>

FS Community,

I created an extension
704
 
I created a
hunt group 7004 that first rings 704 then rings my cell phone 16194548924.
 
I created an
auto attendant 5002 with an option of 704 that goes to hunt group 7004.
 
1)     What does
work: When I call the hunt group directly by dialing 7004 internally, the hunt
group first rings 704 then rings my cell phone successfully.  This is perfect.
 
2)     What does not
work: When I call the main auto attendant from either internally by dialing 5002
or externally by dialing my main number, and dial 704, my extension rings and
after not answering my cell phone does not ring.
 
What I am expecting
in #2 is for my extension 704 to ring and if I do not pick up then my cell
phone will ring just like it does in #1.
 
My knowledge
is limited in this area and if you could point me in the right direction, that
would be great.  I a have tried
everything but with no luck.

Thanks,

-Will



      
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From Mailings at kh-dev.de  Mon Mar  8 14:25:39 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 23:25:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>

Believe it or not...
I heard from people out there who don't have email.
But they use faxing...


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, March 08, 2010 11:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).


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




From javieraristizabal at gmail.com  Mon Mar  8 14:28:53 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:28:53 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
Message-ID: <b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>

Brian. i did with the sip_from_uri, and it is working..

Many thanks

/Javier

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Thanks Brian, let me try your ideas.
>
> /Javier
>
>
> On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian at freeswitch.org> wrote:
>
>> if all else fails try reading do_invite in sofia_glue.c you can really
>> change just about ANYTHING to do with the invite with variables... be
>> careful some let you totally break the packet where it won't even work.  :P
>>
>> /b
>>
>> On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:
>>
>> > Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
>> >
>> > /Javier
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From will.traenkle at yahoo.com  Mon Mar  8 14:31:50 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 14:31:50 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <847524.92329.qm@web57603.mail.re1.yahoo.com>

I am sending this again as I did not see it go through the first time ...


FS Community,


I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

My hunt group code is below:


session.answer();
var domain_name = session.getVariable("domain_name");
var extension = '7000';
var result;
var timeoutpin = 7500;

var objdate = new Date();
var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours
var adjustoperator = "-"; //+ or -
if (adjustoperator == "-") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
if (adjustoperator == "+") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
var Hours = objdate2.getHours();
var Mins = objdate2.getMinutes();
var Seconds = objdate2.getSeconds();
var Month = objdate2.getMonth() + 1;
var Date = objdate2.getDate();
var Year = objdate2.getYear()
var Day = objdate2.getDay()+1;
var exit = false;

 function get_sofia_contact(extension,domain_name, profile){
        if (profile == "auto") {
                        profile = "internal";
                        session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                        sofia_contact = session.getVariable("sofia_contact_"+extension);

                        if (sofia_contact == "error/user_not_registered") {
                                profile = "external";
                                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                                sofia_contact = session.getVariable("sofia_contact_"+extension);
                        }
        }
        else {
                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                sofia_contact = session.getVariable("sofia_contact_"+extension);
        }
        console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" );
        return sofia_contact;
 }


 function mycb( session, type, obj, arg ) {
    try {
        if ( type == "dtmf" ) {
          console_log( "info", "digit: "+obj.digit+"\n" );
          if ( obj.digit == "#" ) {
            //console_log( "info", "detected pound sign.\n" );
            exit = true;
            return( false );
          }

          dtmf.digits += obj.digit;

          if ( dtmf.digits.length >= digitmaxlength ) {
            exit = true;
            return( false );
          }
        }
    } catch (e) {
        console_log( "err", e+"\n" );
    }
    return( true );
 } //end function mycb


dialed_extension = session.getVariable("dialed_extension");
domain_name = session.getVariable("domain_name");
domain = session.getVariable("domain");
us_ring = session.getVariable("us-ring");
caller_id_name = session.getVariable("caller_id_name");
caller_id_number = session.getVariable("caller_id_number");
effective_caller_id_name = session.getVariable("effective_caller_id_name");
effective_caller_id_number = session.getVariable("effective_caller_id_number");
outbound_caller_id_name = session.getVariable("outbound_caller_id_name");
outbound_caller_id_number = session.getVariable("outbound_caller_id_number");
session.execute("set", "ringback=${hold_music}");          //set to ringtone
session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone
session.execute("set", "call_timeout=30");
session.execute("export", "call_timeout=30");
session.execute("set", "continue_on_fail=true");
session.execute("set", "hangup_after_bridge=true");

//console_log( "info", "dialed extension:"+dialed_extension+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "us_ring: "+us_ring+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "domain_name: "+domain_name+".\n" );

//console_log( "info", "action call now don't wait for dtmf\n" );
if ( session.ready() ) {
        session.answer();

        sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto");

        session.execute("bridge", sofia_contact_1000);
        session.execute("bridge", "sofia/gateway/flowroute.com/16194548924");

        //timeout
        if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){
                session.execute("info", "");
                session.execute("transfer", "*991000");
        }

        //clear variables
        dialed_extension = "";
        new_extension = "";
        domain_name = "";
        domain = "";
} //end if session.ready


Thanks.

-Will


      
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From lloyd.aloysius at gmail.com  Mon Mar  8 14:33:33 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 17:33:33 -0500
Subject: [Freeswitch-users] What is the best way implement the follow me for
	a extension
Message-ID: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>

Hi All,

I would like to setup follow me dialplan for individual extensions. Some
extensions need a follow-me and some not. If the the follow-me not reach any
live person the call should go to the FreeSWTICH voicemail.I have the
extensions range from 201 -209. Also I am using the following Default dial
plan to call between extensions.

<extension name="Local_Extension">
      <condition field="destination_number" expression="^(20[0-9])$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML
features"/>
<action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML
features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/>
-->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
<action application="set" data="continue_on_fail=true"/>
<action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
<!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)}"/>-->
<action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@
${domain_name},${followme}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
 </extension>

I use the wiki samples to make a follow me dial plan for extension 201. Here
is the dial plan.

<extension name="201_followme">
  <condition field="destination_number" expression="201">
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="call_timeout=10"/>
   <action application="bridge" data="sofia/$${domain}/201"/>
   <action application="set" data="call_timeout=13"/>
   <action application="bridge" data="sofia/gateway/voicepulse/14161235555"
/>
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default $${domain} 201"/>
  </condition>
</extension>


Call to extension 201 , there are two dialplan matching the request.
Either Local_Extension or follwme_201. Which order freeswtich select?

Thank you for your help.

Thanks,
Lloyd
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From msc at freeswitch.org  Mon Mar  8 14:34:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:34:43 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
Message-ID: <87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:25 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Believe it or not...
> I heard from people out there who don't have email.
> But they use faxing...
>
Whenever you find people like that, ask them if they've ever heard of the
Internet. I hear this Internet thing has a bright future.
-MC
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From brian at freeswitch.org  Mon Mar  8 14:35:19 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:35:19 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
Message-ID: <9F469ABB-1C87-4D11-85D3-FC2B35A1F516@freeswitch.org>

Lets get a search party and go find them and bust that rock they are hiding under.  :P

/b

On Mar 8, 2010, at 4:25 PM, Klaus Hochlehnert wrote:

> Believe it or not...
> I heard from people out there who don't have email.
> But they use faxing...



From msc at freeswitch.org  Mon Mar  8 14:35:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:35:39 -0800
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
	<b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
Message-ID: <87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Brian. i did with the sip_from_uri, and it is working..
>
> Many thanks
>
> /Javier
>

FYI,

I couldn't find this specific tip in the wiki so I added it here:
http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri

-MC
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From brian at freeswitch.org  Mon Mar  8 14:39:08 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:39:08 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
	<87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>
Message-ID: <78F1F8A9-6A65-4F03-BD72-D948CC241781@freeswitch.org>

Don't know about that my parents always told me that computers were a waste of time and the internet was never gonna be a big deal.

So I'm guessing I'm lucky I didn't listen to my parents.

/b

On Mar 8, 2010, at 4:34 PM, Michael Collins wrote:

> Whenever you find people like that, ask them if they've ever heard of the Internet. I hear this Internet thing has a bright future. 
> -MC



From gkuri at ieee.org  Mon Mar  8 14:39:29 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 8 Mar 2010 14:39:29 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>

While I'd love to see FAXing just die, especially with the amount of
headache associated with it, the reality is that specific organizations
still rely on it heavily on a daily basis (ie Health Care and
Banking/Finance Industry). Plus they can legally get away with FAXing
personal data directly between machines, since the FAX transmission doesn't
seem to be directly covered by all their respective privacy regultions (ie
HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
of these organizations just don't have the internal expertise to setup
end-to-end encrypted email nor the impetus to deal with it, since FAXing
works fine in their eyes. I think it's going to be around quite a while
longer unless they actually stop making the FAX machines and force people to
use something else.

Cheers,
Gabe

On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

> Thats unacceptable!!!  Wait its faxing... can't we just let it DIE
> already??  They have this thing called email... you should check it out...
> works great... even lets you attach video, audio and color images.... oh and
> text... dreary old text.... but hey you can mark it all up in HTML and fancy
> fonts now and make it anoying^H^H^H^H^H^H^Hpretty.
>
> /b
>
> On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:
>
> > Together with HylaFAX+ the success rate is ok (unfortunately not
> perfect).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  8 14:46:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:46:44 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>

There is no such thing as security when humans are involved in the relay of information.  Thats fact.

HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are involved its not secure.

/b

On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:

> While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.
> 
> Cheers,
> Gabe



From javieraristizabal at gmail.com  Mon Mar  8 14:47:02 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:47:02 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
	<b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
	<87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>
Message-ID: <b6f11f301003081447r7bc2ab3dy917265f91cef3fd9@mail.gmail.com>

Many Thanks Michael :)

/Javier

On Mon, Mar 8, 2010 at 5:35 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> 2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Brian. i did with the sip_from_uri, and it is working..
>>
>> Many thanks
>>
>> /Javier
>>
>
> FYI,
>
> I couldn't find this specific tip in the wiki so I added it here:
> http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Mailings at kh-dev.de  Mon Mar  8 14:58:39 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 23:58:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>

Me too. I'd also like to see it die.

But honestly I know a lot of small and medium business companies that really rely on faxing.
And for some parts it's much easier than using email. Just say if you get a piece of paper from your tax office and want to send it to your tax consultant.
- Put it in the fax and send it
or
- Scan this piece of paper, save it to the disk, find it (which is btw the most challenging part for most people) and send it by email (most SMB companies don't even have a scanner attached to their PC)

So most people prefer the faxing way... J

Klaus


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Kuri
Sent: Monday, March 08, 2010 11:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

Cheers,
Gabe
On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dave at 3c.co.uk  Mon Mar  8 15:01:04 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 20:01:04 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <FDDF0EE896B34D1A80151B6032B1716C@DELL9>

And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

--Dave
  ----- Original Message ----- 
  From: Gabriel Kuri 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 7:39 PM
  Subject: Re: [Freeswitch-users] Attrafax


  While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

  Cheers,
  Gabe


  On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

    Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

    /b


    On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

    > Together with HylaFAX+ the success rate is ok (unfortunately not perfect).



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





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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Mailings at kh-dev.de  Mon Mar  8 15:02:38 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 00:02:38 +0100
Subject: [Freeswitch-users] Facebook...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>

Hi Guys,

just saw that you already have a Facebook entry.

Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook?
I'd really like to see all news on Facebook...

Cheers,
Klaus
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From brian at freeswitch.org  Mon Mar  8 15:05:56 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 17:05:56 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <FDDF0EE896B34D1A80151B6032B1716C@DELL9>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
Message-ID: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>

Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me...

/b

On Mar 8, 2010, at 5:01 PM, David Knell wrote:

> And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.
>  
> Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.
>  
> --Dave
> 

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From brian at microcomaustralia.com.au  Mon Mar  8 15:08:56 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Tue, 9 Mar 2010 10:08:56 +1100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <3c5cf5261003081508p710c2461g245a688d01f55897@mail.gmail.com>

On 9 March 2010 09:39, Gabriel Kuri <gkuri at ieee.org> wrote:
> Plus they can legally get away with FAXing personal data directly between machines [...]

It is not that uncommon to accidentally send a confidential fax to the
wrong telephone number ... :-(

I have a vague recollection of an incident here where it had been
discovered the government in power had been sending confidential faxes
to a member of the opposition for months until they realized their
records hadn't been updated properly since the last election.

Oops.
-- 
Brian May <brian at microcomaustralia.com.au>


From msc at freeswitch.org  Mon Mar  8 15:10:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 15:10:59 -0800
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
Message-ID: <87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I would like to setup follow me dialplan for individual extensions. Some
> extensions need a follow-me and some not. If the the follow-me not reach any
> live person the call should go to the FreeSWTICH voicemail.I have the
> extensions range from 201 -209. Also I am using the following Default dial
> plan to call between extensions.
>
> <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(20[0-9])$">
> <action application="set" data="dialed_extension=$1"/>
>  <action application="export" data="dialed_extension=$1"/>
> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML
> features"/>
> <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>  <action application="bind_meta_app" data="3 b s execute_extension::cf XML
> features"/>
> <action application="set" data="ringback=${us-ring}"/>
>  <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="set" data="call_timeout=30"/>
>  <!-- <action application="set"
> data="sip_exclude_contact=${network_addr}"/> -->
> <action application="set" data="hangup_after_bridge=true"/>
>  <!--<action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
> -->
>  <action application="set" data="continue_on_fail=true"/>
> <action application="hash"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>  <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
>  <!--<action application="export"
> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
> var sip_secure_media)}"/>-->
>  <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
> <action application="bridge" data="user/${dialed_extension}@
> ${domain_name},${followme}"/>
>  <action application="answer"/>
> <action application="sleep" data="1000"/>
>  <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>       </condition>
>  </extension>
>
> I use the wiki samples to make a follow me dial plan for extension 201.
> Here is the dial plan.
>
> <extension name="201_followme">
>   <condition field="destination_number" expression="201">
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail=true"/>
>    <action application="set" data="ignore_early_media=true"/>
>    <action application="set" data="call_timeout=10"/>
>    <action application="bridge" data="sofia/$${domain}/201"/>
>    <action application="set" data="call_timeout=13"/>
>    <action application="bridge" data="sofia/gateway/voicepulse/14161235555"
> />
>    <action application="answer"/>
>    <action application="sleep" data="1000"/>
>    <action application="voicemail" data="default $${domain} 201"/>
>   </condition>
> </extension>
>
>
> Call to extension 201 , there are two dialplan matching the request.
> Either Local_Extension or follwme_201. Which order freeswtich select?
>

It will match whichever one comes first in the file. You could, for example,
put this 201 extension above the Local_Extension. You could also put a
custom chan var in the user directory for those who need the follow-me. Then
you could have two conditions, both of which need to match:

<extension name="Local, follow me">
  <condition field="${followme}" expression="^yes$"/>
  <condition field="destination_number expression="^(20\d)$">
    <!-- followme stuff here -->
    ...
  </condition>
</extension>

<extension name="Local No follow me">
  <condition field="${followme}" expression="^$"/>
  <condition field="destination_number expression="^(20\d)$">
    <!-- normal local extension stuff here -->
    ...
  </condition>
</extension>

Then in the user setup in the directory add the followme variable only to
those who need it. You can also add a variable for the followme phone number
if you want. Inside the <variables> node just add whatever you want:

  <variable name="followme" value="yes"/>
  <variable name="followme_number" value="14161235555"/>
Or even
  <variable name="followme_dialstring"
value="sofia/gateway/voicepulse/14161235555"/>

Then you could use ${followme_number} in your bridge like this:
  <action application="bridge" data="sofia/gateway/
voicepulse/${followme_number}" />

Or ${followme_dialstring} like this:
  <action application="bridge" data="${followme_dialstring}" />

Make sense? :)

Give it a try and report back your success/failure/questions/comments/etc.
-MC
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From anthony.minessale at gmail.com  Mon Mar  8 15:20:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Mar 2010 17:20:04 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B9562E2.7000207@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
	<4B9562E2.7000207@packetfoundry.net>
Message-ID: <191c3a031003081520y1c2c8bf2r9be0121b28273f0e@mail.gmail.com>

Posting bugs to jira then posting to the mailing list asking us to look at
it is bad etiquette btw.
We already have notification when the bug is posted.


On Mon, Mar 8, 2010 at 2:49 PM, Matt Putnam <matt at packetfoundry.net> wrote:

>  Posted mine up as well.
>
>
> On 3/8/2010 2:33 PM, Brian West wrote:
>
> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
>  And post it to that jira.
>
>  /b
>
>  On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dave at 3c.co.uk  Mon Mar  8 15:21:53 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 20:21:53 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org><8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
Message-ID: <BDEDAB2A5DB04B96B8BD4C3E78DE128E@DELL9>

Quite.  Douglas Adams hit many nails on the head: here's one of them: "We are stuck with technology when what we really want is just stuff that works."

If you want a real challenge, try to fax a document using a WiFi scanner, a Linux box and Pamfax (or pretty much any other online fax service): by the time you get to xsane, you will be..

--Dave

  Me too. I'd also like to see it die.

   

  But honestly I know a lot of small and medium business companies that really rely on faxing.

  And for some parts it's much easier than using email. Just say if you get a piece of paper from your tax office and want to send it to your tax consultant.

  - Put it in the fax and send it

  or

  - Scan this piece of paper, save it to the disk, find it (which is btw the most challenging part for most people) and send it by email (most SMB companies don't even have a scanner attached to their PC)

   

  So most people prefer the faxing way... J

   

  Klaus

   

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Kuri
  Sent: Monday, March 08, 2010 11:39 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Attrafax

   

  While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

  Cheers,
  Gabe

  On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

  Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

  /b


  On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

  > Together with HylaFAX+ the success rate is ok (unfortunately not perfect).



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

   



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gavin.henry at gmail.com  Mon Mar  8 15:22:58 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 23:22:58 +0000
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B9562E2.7000207@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
	<4B9562E2.7000207@packetfoundry.net>
Message-ID: <13ca621c1003081522h422e225brcda68f4b1ecdc76c@mail.gmail.com>

Hi,

That was my bug report. I gave up asking in the #freeswitch channel
and tried to fix it myself. I'm running the debian build from trunk
that you build yourself to produce debs. Mod_spidermoney compiled
correctly and is now running. I forgot to add this info to the JIRA
ticket as that is the best place for code support/issued rather than
irc.

Thanks,

Gavin.

On 08/03/2010, Matt Putnam <matt at packetfoundry.net> wrote:
> Posted mine up as well.
>
> On 3/8/2010 2:33 PM, Brian West wrote:
>> cd /usr/local/freeswitch/mod
>> ldd mod_spidermonkey.so
>>
>> And post it to that jira.
>>
>> /b
>>
>> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>>
>>> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>>>
>>> Has anyone found a solution to the above issue? I am having the same
>>> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>>>
>>>
>>> Thanks,
>>> Matt
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From lloyd.aloysius at gmail.com  Mon Mar  8 15:24:16 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 18:24:16 -0500
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
Message-ID: <8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>

Michael,

Thank you for the suggestion. This suggestion make sense and a proper way to
implement the follow me. .

Thanks
Lloyd

On Mon, Mar 8, 2010 at 6:10 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I would like to setup follow me dialplan for individual extensions. Some
>> extensions need a follow-me and some not. If the the follow-me not reach any
>> live person the call should go to the FreeSWTICH voicemail.I have the
>> extensions range from 201 -209. Also I am using the following Default dial
>> plan to call between extensions.
>>
>> <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(20[0-9])$">
>> <action application="set" data="dialed_extension=$1"/>
>>  <action application="export" data="dialed_extension=$1"/>
>> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>>  <action application="bind_meta_app" data="1 b s execute_extension::dx
>> XML features"/>
>> <action application="bind_meta_app" data="2 b s
>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>  <action application="bind_meta_app" data="3 b s execute_extension::cf
>> XML features"/>
>> <action application="set" data="ringback=${us-ring}"/>
>>  <action application="set" data="transfer_ringback=$${hold_music}"/>
>> <action application="set" data="call_timeout=30"/>
>>  <!-- <action application="set"
>> data="sip_exclude_contact=${network_addr}"/> -->
>> <action application="set" data="hangup_after_bridge=true"/>
>>  <!--<action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
>> -->
>>  <action application="set" data="continue_on_fail=true"/>
>> <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>  <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>> <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>>  <!--<action application="export"
>> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
>> var sip_secure_media)}"/>-->
>>  <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>> <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name},${followme}"/>
>>  <action application="answer"/>
>> <action application="sleep" data="1000"/>
>>  <action application="voicemail" data="default ${domain_name}
>> ${dialed_extension}"/>
>>       </condition>
>>  </extension>
>>
>> I use the wiki samples to make a follow me dial plan for extension 201.
>> Here is the dial plan.
>>
>> <extension name="201_followme">
>>   <condition field="destination_number" expression="201">
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="set" data="continue_on_fail=true"/>
>>    <action application="set" data="ignore_early_media=true"/>
>>    <action application="set" data="call_timeout=10"/>
>>    <action application="bridge" data="sofia/$${domain}/201"/>
>>    <action application="set" data="call_timeout=13"/>
>>    <action application="bridge"
>> data="sofia/gateway/voicepulse/14161235555" />
>>    <action application="answer"/>
>>    <action application="sleep" data="1000"/>
>>    <action application="voicemail" data="default $${domain} 201"/>
>>   </condition>
>> </extension>
>>
>>
>> Call to extension 201 , there are two dialplan matching the request.
>> Either Local_Extension or follwme_201. Which order freeswtich select?
>>
>
> It will match whichever one comes first in the file. You could, for
> example, put this 201 extension above the Local_Extension. You could also
> put a custom chan var in the user directory for those who need the
> follow-me. Then you could have two conditions, both of which need to match:
>
> <extension name="Local, follow me">
>   <condition field="${followme}" expression="^yes$"/>
>   <condition field="destination_number expression="^(20\d)$">
>     <!-- followme stuff here -->
>     ...
>   </condition>
> </extension>
>
> <extension name="Local No follow me">
>   <condition field="${followme}" expression="^$"/>
>   <condition field="destination_number expression="^(20\d)$">
>     <!-- normal local extension stuff here -->
>     ...
>   </condition>
> </extension>
>
> Then in the user setup in the directory add the followme variable only to
> those who need it. You can also add a variable for the followme phone number
> if you want. Inside the <variables> node just add whatever you want:
>
>   <variable name="followme" value="yes"/>
>   <variable name="followme_number" value="14161235555"/>
> Or even
>   <variable name="followme_dialstring"
> value="sofia/gateway/voicepulse/14161235555"/>
>
> Then you could use ${followme_number} in your bridge like this:
>
>   <action application="bridge" data="sofia/gateway/
> voicepulse/${followme_number}" />
>
> Or ${followme_dialstring} like this:
>   <action application="bridge" data="${followme_dialstring}" />
>
> Make sense? :)
>
> Give it a try and report back your success/failure/questions/comments/etc.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar  8 15:28:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 15:28:50 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
Message-ID: <87f2f3b91003081528o6d716414h85b607897b2ab13b@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:58 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Me too. I?d also like to see it die.
>
>
>
> But honestly I know a lot of small and medium business companies that
> really rely on faxing.
>
> And for some parts it?s much easier than using email. Just say if you get a
> piece of paper from your tax office and want to send it to your tax
> consultant.
>
> - Put it in the fax and send it
>
> or
>
> - Scan this piece of paper, save it to the disk, find it (which is btw the
> most challenging part for most people) and
>
Or get a device that will just do the scan & email in one fell swoop. You
could even encrypt before sending and then the person would have to contact
you to get the key before opening it. Not 100% foolproof but it would
prevent the silly, oops, I sent your personal information to the wrong email
address/fax machine. Just think, you could have the convenience of a fax and
relatively secure storage of incoming and outgoing transmissions. Of course,
wanting to cling to 1970's technology is something I can understand. Who
wouldn't want to keep all those Bee Gees and Abba LP's?

-MC
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From freeswitch.org at todandlorna.com  Mon Mar  8 15:30:16 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Mon, 08 Mar 2010 16:30:16 -0700
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
Message-ID: <4B958888.7080902@todandlorna.com>

There exists a large number of companies that make large profits largely 
because they sell based on what they think the customer will pay.  NDAs 
keep customers from comparing prices, and competitors from competing on 
price.  This has been a pretty common telecom practice, though not NDA 
specifically.  This is why the "let me see a copy of your phone bill" 
line is so ill-advised to be substantiated when talking to a new 
provider.  If you give them your phone bill, they know exactly where the 
bar must be set to win.

-Tod Hansmann

On 3/8/2010 4:05 PM, Brian West wrote:
> Why on earth do you need an NDA for prices or any info a provider can 
> send you... sounds pretty lame to me...
>
> /b
>
> On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>> And don't forget the propensity of (wholesale) VoIP providers to 
>> require an NDA to be faxed (or, if you're lucky, scanned and 
>> e-mailed) before discussing pricing.
>> Without wishing to digress further, I find this utterly absurd, and 
>> I've yet to find a satisfactory explanation of why it should be.
>> --Dave
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From freeswitch.org at todandlorna.com  Mon Mar  8 15:30:34 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Mon, 08 Mar 2010 16:30:34 -0700
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
Message-ID: <4B95889A.1090907@todandlorna.com>

There are a number of services that do so.  Implementing them wouldn't 
be trivial, but it would be close.

-Tod Hansmann

On 3/8/2010 4:02 PM, Klaus Hochlehnert wrote:
>
> Hi Guys,
>
> just saw that you already have a Facebook entry.
>
> Wouldn't it be possible to connect the news/twitter feeds somehow with 
> Facebook?
>
> I'd really like to see all news on Facebook...
>
> Cheers,
>
> Klaus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From brian at freeswitch.org  Mon Mar  8 15:34:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 17:34:00 -0600
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <4B95889A.1090907@todandlorna.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
Message-ID: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>

Would love to have someone step in and help with these types of things.  As it stands I fight for the time to sleep.  :P

/b

On Mar 8, 2010, at 5:30 PM, Tod Hansmann wrote:

> There are a number of services that do so.  Implementing them wouldn't be trivial, but it would be close.
> 
> -Tod Hansmann



From Mailings at kh-dev.de  Mon Mar  8 15:48:06 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 00:48:06 +0100
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
	<5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4B@srv02.khdev.corp>

I just found a (German) help on how to do this.
Looks like there's a Facebook app that can do this
http://help.twitter.com/entries/86007-erste-schritte-twitter-auf-facebook

If it's as easy as they describe I could help.
I've never done this before, but I'm willing to learn... ;-)


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, March 09, 2010 12:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Facebook...

Would love to have someone step in and help with these types of things.  As it stands I fight for the time to sleep.  :P

/b

On Mar 8, 2010, at 5:30 PM, Tod Hansmann wrote:

> There are a number of services that do so.  Implementing them wouldn't be trivial, but it would be close.
>
> -Tod Hansmann


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




From anthony.minessale at gmail.com  Mon Mar  8 15:57:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Mar 2010 17:57:25 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4B958888.7080902@todandlorna.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
Message-ID: <191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>

All these threads seem to go off on on tangents, seems at least a lot of
people are reading it so.....

This thread brought to you by ClueCon MMX!
register today http://www.cluecon.com

Hey, it was worth a shot, maybe we can do a presentation on t38 in FS (if we
ever get the time to finish it).




On Mon, Mar 8, 2010 at 5:30 PM, Tod Hansmann <freeswitch.org at todandlorna.com
> wrote:

>  There exists a large number of companies that make large profits largely
> because they sell based on what they think the customer will pay.  NDAs keep
> customers from comparing prices, and competitors from competing on price.
> This has been a pretty common telecom practice, though not NDA
> specifically.  This is why the "let me see a copy of your phone bill" line
> is so ill-advised to be substantiated when talking to a new provider.  If
> you give them your phone bill, they know exactly where the bar must be set
> to win.
>
> -Tod Hansmann
>
>
> On 3/8/2010 4:05 PM, Brian West wrote:
>
> Why on earth do you need an NDA for prices or any info a provider can send
> you... sounds pretty lame to me...
>
>  /b
>
>  On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>  And don't forget the propensity of (wholesale) VoIP providers to require
> an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before
> discussing pricing.
>
> Without wishing to digress further, I find this utterly absurd, and I've
> yet to find a satisfactory explanation of why it should be.
>
> --Dave
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From steveu at coppice.org  Mon Mar  8 16:34:36 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 09 Mar 2010 08:34:36 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>	<4B951E63.6060200@coppice.org>	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
Message-ID: <4B95979C.7070206@coppice.org>

On 03/09/2010 05:54 AM, Klaus Hochlehnert wrote:
> Hi
>
> you could also use newest trunk of t38modem + trunk of OPAL.
> The newest trunk version(s) use SpanDSP as fax engine and it works more reliable than the previous t38 engine (I think it was integrated in OPAL).
> It also supports fallback to t30 audio fax if you're connected to ISDN network.
>
> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).
>
> Klaus
>    
The success rate with the latest opal + spandsp code should be about as 
about good as your system is capable of giving. That is, most failures 
will either be:

     - that the call is just a bad call - wrong number, messed up FAX 
machine usage, etc.
     - your systems is dropping packets or otherwise screwing up the timing.
     - T.38 negotiation is screwing up, because T.38 compatibility is 
still a mess.

The last one is such a problem for the whole industry, the SIP Forum has 
a working group to address it.

Steve



From steveu at coppice.org  Mon Mar  8 16:38:02 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 09 Mar 2010 08:38:02 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
Message-ID: <4B95986A.4020708@coppice.org>

On 03/09/2010 07:05 AM, Brian West wrote:
> Why on earth do you need an NDA for prices or any info a provider can 
> send you... sounds pretty lame to me...
>
> /b
Lame? The essence of effective negotiating is to keep all the parties 
involved divided, and each thinking they got a great deal.

"Don't tell the others about this price offer, will you" tends to mean 
"if you talk about these prices, you might find others are paying less".

Steve

>
> On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>> And don't forget the propensity of (wholesale) VoIP providers to 
>> require an NDA to be faxed (or, if you're lucky, scanned and 
>> e-mailed) before discussing pricing.
>> Without wishing to digress further, I find this utterly absurd, and 
>> I've yet to find a satisfactory explanation of why it should be.
>> --Dave
>>



From will.traenkle at yahoo.com  Mon Mar  8 16:45:10 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 16:45:10 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <304682.93065.qm@web57605.mail.re1.yahoo.com>

I did not see this message go through yet so I removed the code I originally had pasted and am resending it.

FS Community,



I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

Thanks.

-Will


      
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From msc at freeswitch.org  Mon Mar  8 16:55:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 16:55:28 -0800
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <304682.93065.qm@web57605.mail.re1.yahoo.com>
References: <304682.93065.qm@web57605.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003081655o5e1adf0ap868ec31b0631aab6@mail.gmail.com>

I thought someone responded already and asked that you confirm that you are
ignoring early media in your bridge line...

-MC

On Mon, Mar 8, 2010 at 4:45 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> I did not see this message go through yet so I removed the code I
> originally had pasted and am resending it.
>
> FS Community,
>
> I am new to freeSWITCH and this mailing list and I appreciate your support
> in advance.
>
> My Goal: To create a hunt group x7000 that sequentially dials an internal
> extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while
> playing music the entire time when each number is being tried.
>
> The Issue: This was working great a few months back, but now music plays
> when only when x1000 is tried but NOT when the cell phone is tried.
>
> Ideas? Please advise.
>
> Thanks.
>
> -Will
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Mar  8 17:01:42 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Mar 2010 09:01:42 +0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
	<23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
	<5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>
Message-ID: <23f91031003081701h4841f7e7x69def77879707117@mail.gmail.com>

The patch was committed on 15433. Though I haven't start to use that.
Which version are you on?
http://jira.freeswitch.org/browse/MODLANG-122

2010/3/9 Mike van Lammeren <mike at van.lammeren.net>:
> Actually, I don't believe that does work. At least, not on my system. Trying
> to access argv[0] just produces a lua error about "concatenating a nil
> value."
>
> On Fri, Mar 5, 2010 at 10:28 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> and also you can get the script name from argv[0] if you name you
>> script in different names
>>
>> 2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
>> > In case anyone's interested, I just found something cool. You can pass
>> > parameters to the Lua script from the lua.conf.xml file, like so:
>> > ?? ?<param name="startup-script" value="temp.lua script1"/>
>> > ?? ?<param name="startup-script" value="temp.lua script2"/>
>> > ?? ?<param name="startup-script" value="temp.lua script3"/>
>> > I can see that being useful.
>> > Mike van Lammeren
>> >
>> > On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren
>> > <mike at van.lammeren.net>
>> > wrote:
>> >>
>> >> Thanks so much, everyone! Awesome advice, as usual!
>> >> Due to a deadline situation, I'm going with the Lua script idea.
>> >> However,
>> >> if it gives us any trouble after the looming deadline, then I will
>> >> definitely go the C route.
>> >> Thanks again!
>> >> Mike van Lammeren
>> >>
>> >> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org>
>> >> wrote:
>> >>>
>> >>> Write it in C and use the FreeSWITCH task scheduler API.
>> >>>
>> >>> /b
>> >>>
>> >>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>> >>>
>> >>> > All the same, I'm still open to suggestions on better ways to
>> >>> > achieve
>> >>> > my ultimate goals!
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From null at invalid.name  Mon Mar  8 17:05:34 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 9 Mar 2010 01:05:34 +0000
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
	<5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
Message-ID: <b042b23b1003081705r26bf624ep82e732e328cb30f6@mail.gmail.com>

On Mon, Mar 8, 2010 at 11:34 PM, Brian West <brian at freeswitch.org> wrote:
> Would love to have someone step in and help with these types of things. ?As it stands I fight for the time to sleep. ?:P

I could probably spare some time each week looking after social media
/ news stuff for freeswitch if required.

ISTR I've got you on Jabber (null at invalid.name) so grab me off-list to discuss.


From brian at freeswitch.org  Mon Mar  8 17:07:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 19:07:55 -0600
Subject: [Freeswitch-users] SILK Codec
Message-ID: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>

We now have mod_silk in tree it has only been tested on Linux so far.  The silk library itself was released in the IETF draft only an hour and twenty minutes ago.  I wrote a perl script to extract the source from the draft.  Checked in the lib and the codec module which I had written to the binary lib I had a few months ago. 

It has more work to do... but its there if anyone wants to chip in and libtoolize the library, revamp the build system and assist in testing the codec on multiple platforms.

Thanks,
Brian



From dave at 3c.co.uk  Mon Mar  8 17:20:03 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 22:20:03 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9><2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
Message-ID: <90C16E97D2234FD58E5099C04329ACC3@DELL9>

Hi Tod,

You might be right in certain circumstances, but, in wholesale telecoms, it's pretty easy to get a good idea of where rates for particular destinations are.  Find a collection of ratecards by typing 'wholesale sip termination' into Google and it quickly becomes pretty obvious what's competitive and what isn't.  NDAs are completely superfluous, especially considering that so much buying is done on price.  If I've not got a good reason to want to terminate traffic through you (i.e. you're not AT&T or Verizon or the like), then I'm unlikely to want to go to the trouble of signing an NDA just to see how much your product costs.

--Dave
  ----- Original Message ----- 
  From: Tod Hansmann 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 8:30 PM
  Subject: Re: [Freeswitch-users] Attrafax


  There exists a large number of companies that make large profits largely because they sell based on what they think the customer will pay.  NDAs keep customers from comparing prices, and competitors from competing on price.  This has been a pretty common telecom practice, though not NDA specifically.  This is why the "let me see a copy of your phone bill" line is so ill-advised to be substantiated when talking to a new provider.  If you give them your phone bill, they know exactly where the bar must be set to win.

  -Tod Hansmann

  On 3/8/2010 4:05 PM, Brian West wrote: 
    Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me... 


    /b


    On Mar 8, 2010, at 5:01 PM, David Knell wrote:


      And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

      Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

      --Dave





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

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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jaybinks at gmail.com  Mon Mar  8 17:27:57 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 11:27:57 +1000
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
Message-ID: <ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>

this is what I love about Freeswitch, and the FS development team.

Well done Brian ( and the rest of the team ).

Jay



On Tue, Mar 9, 2010 at 11:07 AM, Brian West <brian at freeswitch.org> wrote:

> We now have mod_silk in tree it has only been tested on Linux so far.  The
> silk library itself was released in the IETF draft only an hour and twenty
> minutes ago.  I wrote a perl script to extract the source from the draft.
>  Checked in the lib and the codec module which I had written to the binary
> lib I had a few months ago.
>
> It has more work to do... but its there if anyone wants to chip in and
> libtoolize the library, revamp the build system and assist in testing the
> codec on multiple platforms.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From infos at madovsky.org  Mon Mar  8 17:45:54 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 20:45:54 -0500
Subject: [Freeswitch-users] create phone user extension
Message-ID: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>

to create a  phone user extension,
is it need to have his account in conf/directory/default for default.xml ?
if yes so I don't understand why I can register with extension 1003 if there are only 1000-1002 in conf/directory
I use the defaults FS config

Thanks
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From mayamatakeshi at gmail.com  Mon Mar  8 17:59:58 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 9 Mar 2010 10:59:58 +0900
Subject: [Freeswitch-users] create phone user extension
In-Reply-To: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
Message-ID: <15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>

On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:

>  to create a  phone user extension,
> is it need to have his account in conf/directory/default for default.xml ?
>

Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve
directory info.


> if yes so I don't understand why I can register with extension 1003 if
> there are only 1000-1002 in conf/directory
> I use the defaults FS config
>
>
Is the auth-calls parameter set to true in your sofia profile?
<param name="auth-calls" value="true"/>
If not, anyone can register.
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From infos at madovsky.org  Mon Mar  8 18:22:38 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 21:22:38 -0500
Subject: [Freeswitch-users] create phone user extension
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
	<15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>
Message-ID: <338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>

auth-calls are exactly as default install config
<param name="auth-calls" value="$${internal_auth_calls}"/>

for now I have 2 freeeswitch with pgsql DB shared. so does it use mod_xml_odbc implicitly ?
  ----- Original Message ----- 
  From: mayamatakeshi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 8:59 PM
  Subject: Re: [Freeswitch-users] create phone user extension




  On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:

    to create a  phone user extension,
    is it need to have his account in conf/directory/default for default.xml ?

  Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve directory info.
   

    if yes so I don't understand why I can register with extension 1003 if there are only 1000-1002 in conf/directory
    I use the defaults FS config



  Is the auth-calls parameter set to true in your sofia profile?
  <param name="auth-calls" value="true"/>
  If not, anyone can register.



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch at gilligan.id.au  Mon Mar  8 18:30:34 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 13:30:34 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com> 
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
Message-ID: <9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>

That decently woks to originate the call but passing that UUID into a new
ManagedSession does not seem to like to the session created using that UUID.
 Once i create a session used the API you suggested how would i then access
the session in the Managed Code?

Chris

On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> You could use the api:
>
>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
> string apiResult = string.Empty;
> string Uuid;
> string NumberToDial = "3475558308";
> string OutgoingCallerID = "2155556666";
>
> Uuid = fsApi.ExecuteString("create_uuid");
>
> apiResult = fsApi.Execute("originate",
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
> Uuid, OutgoingCallerID, NumberToDial));
>
>
>
>
>
>
> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Hi,
>> I am trying to create a mod_managed API application that takes 2 phone
>> numbers as params.  These numbers should then be used to make 2 calls and to
>> bridge them. The issue i am having is working out how to place the first
>> call leg from in mod_managed.  All the methods i have found are for the
>> second leg and require you to pass in the first call leg.
>>
>> I know there are easier way to do this out of the managed code but i want
>> it in the managed code as this is just a proof of concept to prove we can do
>> a callback like system via managed code to be hooked into other external
>> systems.
>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>
>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>> while (s.ready()) {
>> // The call has been answered
>> }
>>
>> This seems to be exactly what i am looking for but seems to be missing
>> from mod_managed. I am hoping someone can tell me how to do something
>> similar in mod_managed since even if it is not part of the managed code
>> mod_managed is meant to have the native access as well so i would assume it
>> would be possible.
>>
>> Thanks in advance
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Mon Mar  8 18:48:08 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 9 Mar 2010 11:48:08 +0900
Subject: [Freeswitch-users] create phone user extension
In-Reply-To: <338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
	<15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>
	<338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>
Message-ID: <15b9404e1003081848q39332c81s5c7a4c1e4f84fa91@mail.gmail.com>

On Tue, Mar 9, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

>  auth-calls are exactly as default install config
> <param name="auth-calls" value="$${internal_auth_calls}"/>
>

Take a look at
/usr/local/freeswitch/log/freeswitch.xml.fsxml
and check what is the value it was resolved at run-time.


> for now I have 2 freeeswitch with pgsql DB shared. so does it use
> mod_xml_odbc implicitly ?
>

No. I just said that directory is not restricted to putting files at
conf/directory
and that you you could use mod_xml_odbc or mod_xml_curl.
But if you are using default FS files, then you are not using them.



>  ----- Original Message -----
> *From:* mayamatakeshi <mayamatakeshi at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 08, 2010 8:59 PM
> *Subject:* Re: [Freeswitch-users] create phone user extension
>
>
> On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  to create a  phone user extension,
>> is it need to have his account in conf/directory/default for default.xml ?
>>
>
> Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve
> directory info.
>
>
>>  if yes so I don't understand why I can register with extension 1003 if
>> there are only 1000-1002 in conf/directory
>> I use the defaults FS config
>>
>>
> Is the auth-calls parameter set to true in your sofia profile?
> <param name="auth-calls" value="true"/>
> If not, anyone can register.
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nagalenoj at gmail.com  Mon Mar  8 20:33:15 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Mar 2010 10:03:15 +0530
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
In-Reply-To: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
References: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
Message-ID: <fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>

Anything wrong with this question?! I didn't find any replies for this..

On Fri, Mar 5, 2010 at 10:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>      I've faced an issue in event socket. I would want to know why it
> behaves such a way.
>
>      My program is working with the hep of events. So, based on the
> received event, the process will continue it's work. When I need DTMF, I use
> recvEventTimed and in the other cases, I use recvEvent.
>      So, In the mid if caller hangsup, I expect for the SERVER_DISCONNECTED
> event. When the caller hangsup when I'm waiting in recvEvent, I'm getting
> SERVER_DISCONNECTED. But, when I'm waiting in recvEventTimed, I'mnot
> receiving SERVER_DISCONNECTED, instead receiving an undefined value.
>
>     To handle this, I've checked esl connection inside the timeout part and
> put a recvEvent, then I receive SERVER_DISCONNECTED.
>
>     My question is, why am I not receiving SERVER_DISCONNECTED when I recv
> event using recvEventTimed?!
>
> This is a sample program I execute,
>
> require ESL;
> use IO::Socket::INET;
>
> my $ip = "127.0.0.1";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '9242',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
>
> for(;;) {
>   my $new_sock = $sock->accept();
>   my $pid = fork();
>   if ($pid) {
>     close($new_sock);
>     next;
>   }
>
>   my $host = $new_sock->sockhost();
>   my $fd = fileno($new_sock);
>
>   my $con = new ESL::ESLconnection($fd);
>   my $info = $con->getInfo();
>   my $uuid = $info->getHeader("unique-id");
>   printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>
>   my $e = $con->filter("unique-id", $uuid);
>   if ($e) {
>     print $e->serialize();
>   } else {
>     printf("WTF?\n");
>   }
>
>   $con->events("plain", "SERVER_DISCONNECTED DTMF");
>   $con->execute("answer");
>
>   while($con->connected()) {
>     $e = $con->recvEvent();   ############# CASE 1 ##
>     # $e = $con->recvEventTimed(10000);       ############# CASE 2 ##
>
>     unless ($e) {
>         unless ($con->connected()) {
> ###############################
>            $e = $con->recvEvent();                         ##
>            print $e->serialize();                               ##   Added
> code to get SERVER_DISCONNECTED.
>            print "SERVER_DISCONNECTED";     ##
>         }
> ###############################
>         print "DTMF timeout\n";
>     }
>     if ($e) {
>       my $name = $e->getHeader("event-name");
>       print "EVENT [$name]\n";
>       if ($name eq "DTMF") {
>         my $digit = $e->getHeader("dtmf-digit");
>         my $duration = $e->getHeader("dtmf-duration");
>         print "DTMF digit $digit ($duration)\n";
>       }
>     }
>   }
>   print "BYE\n";
>   close($new_sock);
> }
>
> --
> Regards,
> Nagalenoj H.
>



-- 
Regards,
Nagalenoj H.
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From brian at freeswitch.org  Mon Mar  8 20:49:02 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 22:49:02 -0600
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
In-Reply-To: <fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>
References: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
	<fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>
Message-ID: <D19C6489-26D8-4AD3-A5EF-E2D6436A13F6@freeswitch.org>

This is because you're timing out and ending your loop and loosing the event.   while($con->connected()) { becomes false and you can't come back around to pickup the event because you would have been disconnected before you had a chance to get the event.   recvEventTimed allows you to do other things if you don't receive events in a set amount of time.

One example of proper usage on inbound event socket:

http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/ESL/Dispatch.pm
http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/dispatch.pl


/b
On Mar 8, 2010, at 10:33 PM, Nagalenoj H. wrote:

> Anything wrong with this question?! I didn't find any replies for this..



From infos at madovsky.org  Mon Mar  8 21:58:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 00:58:10 -0500
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
Message-ID: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>

I noticed that if a registration doesn't close properly and
try to reconnect I can see in the table sip_registrations the old reg and new one.
is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?

Thanks

Franck
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From brian at freeswitch.org  Mon Mar  8 22:13:24 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 00:13:24 -0600
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
In-Reply-To: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
References: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
Message-ID: <76633F3B-93B1-47CF-A3E8-FDD07FD817D3@freeswitch.org>

old reg? what do you mean and how do you mean ?  and the hostname is how it tells.
On Mar 8, 2010, at 11:58 PM, Madovsky wrote:

> I noticed that if a registration doesn't close properly and
> try to reconnect I can see in the table sip_registrations the old reg and new one.
> is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?
>  
> Thanks
>  
> Franck
> _______

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From jbrucehopkins at gmail.com  Mon Mar  8 22:22:52 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Mar 2010 06:22:52 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
Message-ID: <63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>

Ah I see.  I will try again using the internal profile and forwarding  
port 5060.  Presumably still creating a directory entry to enable the  
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two  
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing  
>> something wrong here as I am still having difficulty, and still  
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding  
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to  
>> retain an entry in the directory /usr/local/freeeswitch/conf/ 
>> directory/ext_dns.xml in order to give freeswitch the dns name of  
>> the server as a domain for the remote softphone to register on.  I  
>> left the group name in this entry the same as inthe default entry,  
>> so that the remote phone could register on the same extension  
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN  
>> as freeswitch) phone to the remote phone, I get the message on the  
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of  
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment  
>> <param name="aggressive-nat-detection" value="true"/> in  
>> external.xml, otherwise in the case of a call initiated by the  
>> remote phone being hung up by the local phone, freeswitch sent the  
>> BYE to the private IP of the remote phone, rather than its public  
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From infos at madovsky.org  Mon Mar  8 22:29:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 01:29:18 -0500
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
References: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
	<76633F3B-93B1-47CF-A3E8-FDD07FD817D3@freeswitch.org>
Message-ID: <05F13F0A7E3B40C78F8BB4BC1061032A@MOBILEE1705>

I mean if legA register on host1, suddenly FS or the client shutdown and register again on host1 or host2 so
in pgsql the old call_id row exsits yet, and a new row with the ne call_id  is created...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:13 AM
  Subject: Re: [Freeswitch-users] sip registrations and postgresql odbc-dsn


  old reg? what do you mean and how do you mean ?  and the hostname is how it tells.

  On Mar 8, 2010, at 11:58 PM, Madovsky wrote:


    I noticed that if a registration doesn't close properly and
    try to reconnect I can see in the table sip_registrations the old reg and new one.
    is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?

    Thanks

    Franck
    _______




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


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

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

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From infos at madovsky.org  Mon Mar  8 23:59:40 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 02:59:40 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
Message-ID: <663CE110CAE64F77BE2FEF42B1905A2D@MOBILEE1705>

in sip_registration table there are
call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname 
fields, so how can I get these vars from the callee before the bridge ?
I need to replace the host of the callee in case of if he's registered on another FS node.

thanks

Franck
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From infos at madovsky.org  Tue Mar  9 00:04:22 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 03:04:22 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
Message-ID: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>

forgot to say that I need it (since I didn't find any good examples)
for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:59 AM
  Subject: vars from sip_registrations table


  in sip_registration table there are
  call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname 
  fields, so how can I get these vars from the callee before the bridge ?
  I need to replace the host of the callee in case of if he's registered on another FS node.

  thanks

  Franck
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From will.traenkle at yahoo.com  Tue Mar  9 00:47:50 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 00:47:50 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <20100308184025.GL18427@base.carmickle.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
	<20100308184025.GL18427@base.carmickle.com>
Message-ID: <358354.17194.qm@web57606.mail.re1.yahoo.com>

Thanks Frank.  I will try that out.

Cheers,




________________________________
From: Frank Carmickle <frank at carmickle.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, March 8, 2010 10:40:25 AM
Subject: Re: [Freeswitch-users] hunt group - play music when trying external numbers

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC

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



      
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From siniypin at gmail.com  Tue Mar  9 01:17:23 2010
From: siniypin at gmail.com (RobertT)
Date: Tue, 9 Mar 2010 12:17:23 +0300
Subject: [Freeswitch-users] ICE related sdp headers forwarding
Message-ID: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>

Hi guys!

In my installation I have a public FS being connected by lots of clients and
being used to connect them with each other. Some of them might be backed by
various NATs. And though they all use STUN to discover their public IP
address, sometimes it's just not enough to provide two-way sound. These
clients all have ICE and TURN support, but it is useless, since FS doesn't
support ICE (at least that's what I know), and there are no ICE-related
headers in FS's originating SDP for b-leg. Currently, all calls are being
connected by means of bridge app with proxy_media mode enabled.

For now, I can only think of couple of ways to enable ICE negotiation
between a and b:
1. Redirect a directly to b. But to accomplish this I've to know b's IP,
that it was registered with, which is being stored in sip_registrations
db. Can I obtain it with db app in dialplan and substitute as a parameter to
deflect app?
2. Force FS to copy remote SDP headers, or at least a part of them, to it's
local SDP.
3. I feel I might be missing something important...

Please help me to find a best solution to enable ICE negotiation.

Best regards, RobertT.
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From yehavi.bourvine at gmail.com  Tue Mar  9 01:45:24 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 11:45:24 +0200
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
Message-ID: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>

Is there a simple way/command inside FreeSwitch to send a NOTIFY message to
a Polycom phone which tells it to reboot? I know it is possible but I do not
find any documentation about it...

                     Thanks, __Yehavi:
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From jaybinks at gmail.com  Tue Mar  9 02:08:13 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 20:08:13 +1000
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
In-Reply-To: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
References: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
Message-ID: <ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>

this should help you on your way.

http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints

<http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints>
Jay

On Tue, Mar 9, 2010 at 7:45 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Is there a simple way/command inside FreeSwitch to send a NOTIFY message to
> a Polycom phone which tells it to reboot? I know it is possible but I do not
> find any documentation about it...
>
>                      Thanks, __Yehavi:
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From freeswitch at gilligan.id.au  Tue Mar  9 02:27:31 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 21:27:31 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
Message-ID: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>

What i need to do is listen on a session before, during and after the
session is bridged with another call. I know built into FS is the option to
listen for *x where x is 0-9 but i really need more than one digit.

I traced the method that listens for the single digit to the
command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
switch_ivr_async.c

mod_managed also has the following in the native class freeswitch.
public static switch_status_t
switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
recv_dtmf)

The problem is i can't work out how to use it.  I am fine the the first
param for the session but have no idea what i need for
the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
recv_dtmf param.

if anyone can point me in the right direction or knows a better way i would
be grateful.

Chris
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From yehavi.bourvine at gmail.com  Tue Mar  9 03:13:16 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 13:13:16 +0200
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
In-Reply-To: <ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>
References: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
	<ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>
Message-ID: <d361ca7b1003090313x217e7ac8n686a687285a127d6@mail.gmail.com>

Thanks!
                 __Yehavi:

 2010/3/9 jay binks <jaybinks at gmail.com>

> this should help you on your way.
>
>
> http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints
>
> <http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints>
> Jay
>
>  On Tue, Mar 9, 2010 at 7:45 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Is there a simple way/command inside FreeSwitch to send a NOTIFY message
>> to a Polycom phone which tells it to reboot? I know it is possible but I do
>> not find any documentation about it...
>>
>>                      Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Stephen.Kingham at kingtech.com.au  Tue Mar  9 00:18:39 2010
From: Stephen.Kingham at kingtech.com.au (Stephen Kingham)
Date: Tue, 09 Mar 2010 19:18:39 +1100
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <4B9532AB.5030407@telemaque.fr>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr>
Message-ID: <1268122719.27572.2.camel@zen>

Hi

I can help out with the Cisco config, but I have not worked out what I
need to configure FreeSWITCH to route a SIP call to the ip address of
the Cisco GW.  If I could get a bit of help there I will write, or help
write howto.

Regards

Stevek


On Mon, 2010-03-08 at 18:23 +0100, Tristan Mah? wrote:
> +1 with Brian,
> 
> Just as I have 5 spare minutes and to be kind:
> 
> 
> * E1 controller config:
> 
> controller E1 0/0/0
>  framing NO-CRC4
>  pri-group timeslots 1-31
>  description E1_1
> !
> interface Serial0/0/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-net5
>  isdn not-end-to-end 64
>  isdn incoming-voice voice
>  isdn bchan-number-order descending round-robin
>  no cdp enable
> !
> voice-port 0/0/0:15
>  no comfort-noise
>  cptone FR
>  timeouts call-disconnect 1
>  timeouts wait-release 1
> !
> 
> * Dial-peers ( SIP to E1 and E1 to SIP):
> 
> dial-peer voice 100 pots
>  description SIP_TO_E1
>  destination-pattern 0........
>  progress_ind setup enable 3
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  progress_ind disconnect enable 8
>  no digit-strip
>  port 0/0/0:15
> !
> dial-peer voice 500 voip
>  description E1_TO_SIP
>  destination-pattern ....
>  progress_ind setup enable 3
>  session protocol sipv2
>  session target ipv4:FREESWITCH_IP
>  session transport udp
>  dtmf-relay rtp-nte
>  codec g711alaw
>  no vad
> !
> 
> 
> Remember to adapt this config to your needs ( there are mandatory
> missing parts, but at least you have some config options to look for).
> 
> Regards,
> 
> Gled.
> 
> Brian West a ?crit :
> > You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.
> > 
> > /b
> > 
> > On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> > 
> >> HI,
> >>  
> >> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
> >>  
> >> Thanks
> >> Srinivas
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jaybinks at gmail.com  Tue Mar  9 04:47:05 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 22:47:05 +1000
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <1268122719.27572.2.camel@zen>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com> 
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com> 
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr> <1268122719.27572.2.camel@zen>
Message-ID: <ec2ce17e1003090447l78eef33x643e84289d3fb2d8@mail.gmail.com>

all you need is..

<action application="bridge"
data="{sip_cid_type=rpid}sofia/**YOURPROFILE**/$1@**CISCOIP**" />

not all IOS's are going to need the caller ID to be forced to rpid,
but in my experience I needed it.  ( it should not hurt anyways )

hope that helps

Jay

On Tue, Mar 9, 2010 at 6:18 PM, Stephen Kingham <
Stephen.Kingham at kingtech.com.au> wrote:

> Hi
>
> I can help out with the Cisco config, but I have not worked out what I
> need to configure FreeSWITCH to route a SIP call to the ip address of
> the Cisco GW.  If I could get a bit of help there I will write, or help
> write howto.
>
> Regards
>
> Stevek
>
>
> On Mon, 2010-03-08 at 18:23 +0100, Tristan Mah? wrote:
> > +1 with Brian,
> >
> > Just as I have 5 spare minutes and to be kind:
> >
> >
> > * E1 controller config:
> >
> > controller E1 0/0/0
> >  framing NO-CRC4
> >  pri-group timeslots 1-31
> >  description E1_1
> > !
> > interface Serial0/0/0:15
> >  no ip address
> >  encapsulation hdlc
> >  isdn switch-type primary-net5
> >  isdn not-end-to-end 64
> >  isdn incoming-voice voice
> >  isdn bchan-number-order descending round-robin
> >  no cdp enable
> > !
> > voice-port 0/0/0:15
> >  no comfort-noise
> >  cptone FR
> >  timeouts call-disconnect 1
> >  timeouts wait-release 1
> > !
> >
> > * Dial-peers ( SIP to E1 and E1 to SIP):
> >
> > dial-peer voice 100 pots
> >  description SIP_TO_E1
> >  destination-pattern 0........
> >  progress_ind setup enable 3
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  progress_ind disconnect enable 8
> >  no digit-strip
> >  port 0/0/0:15
> > !
> > dial-peer voice 500 voip
> >  description E1_TO_SIP
> >  destination-pattern ....
> >  progress_ind setup enable 3
> >  session protocol sipv2
> >  session target ipv4:FREESWITCH_IP
> >  session transport udp
> >  dtmf-relay rtp-nte
> >  codec g711alaw
> >  no vad
> > !
> >
> >
> > Remember to adapt this config to your needs ( there are mandatory
> > missing parts, but at least you have some config options to look for).
> >
> > Regards,
> >
> > Gled.
> >
> > Brian West a ?crit :
> > > You seem rather needy.  Have you even tried google yet?  When asking
> for help you could be more descriptive if you want help.  These "I need
> help" without any directed questioned signals to me you have a job to do and
> you don't know how to do it and you're now asking US to do your job for you.
>   Be descriptive and do some research yourself.  Again we aren't cisco and
> you didn't pay us for that hardware so I would start by venturing over to
> www.cisco.com and searching their docs for that hardware on how to
> configure voice peers.  Then ask questions if anything is left to question.
> > >
> > > /b
> > >
> > > On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> > >
> > >> HI,
> > >>
> > >> how do i configure freeswitch gateway to communicate to cisco router,
> i am using cisco2921.
> > >>
> > >> Thanks
> > >> Srinivas
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From srinivas.ksvreddy at gmail.com  Tue Mar  9 05:47:23 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 9 Mar 2010 19:17:23 +0530
Subject: [Freeswitch-users] freeswitch to cisco router
Message-ID: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>

Hi,

i have tried to configure gateway from my freeswitch, with the following
credentials proxy(ip of cisco) and username,password.
and when i start my freeswitch registration reqest is going to cisco router
and again it requesting to same registration reqeust to my sipserver, the
register packet from cisco router to my freeswitch is comming like this

To: sip:.*@sipserverip:5060

so there there is no user present with the name of .* it is rejecting the
registration request, any idea

Thanks
Swapna
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From yehavi.bourvine at gmail.com  Tue Mar  9 06:24:39 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 16:24:39 +0200
Subject: [Freeswitch-users] freeswitch to cisco router
In-Reply-To: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
References: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
Message-ID: <d361ca7b1003090624p202563e2k434d9aabe7a26559@mail.gmail.com>

Cisco gateways usually do not register with the server (thus you have to
allow them via ACL). If you post your config we might help.

                 __Yehavi:

 2010/3/9 srinivasula reddy <srinivas.ksvreddy at gmail.com>

>  Hi,
>
> i have tried to configure gateway from my freeswitch, with the following
> credentials proxy(ip of cisco) and username,password.
> and when i start my freeswitch registration reqest is going to cisco router
> and again it requesting to same registration reqeust to my sipserver, the
> register packet from cisco router to my freeswitch is comming like this
>
> To: sip:.*@sipserverip:5060
>
> so there there is no user present with the name of .* it is rejecting the
> registration request, any idea
>
> Thanks
> Swapna
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Tue Mar  9 07:43:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 09:43:10 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B955401.1030004@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
	<4B955401.1030004@marketelsystems.com>
Message-ID: <4B966C8E.9040805@marketelsystems.com>

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From anthony.minessale at gmail.com  Tue Mar  9 08:39:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 10:39:17 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
Message-ID: <191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>

You would need to make a patch for it or pay someone to add it.


On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

>  forgot to say that I need it (since I didn't find any good examples)
> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
> not compatible)
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 2:59 AM
> *Subject:* vars from sip_registrations table
>
> in sip_registration table there are
> call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>
> fields, so how can I get these vars from the callee before the bridge ?
> I need to replace the host of the callee in case of if he's registered on
> another FS node.
>
> thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From infos at madovsky.org  Tue Mar  9 08:47:35 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 11:47:35 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <F996975CC7C34E28A54D55CEBC5E4CB0@MOBILEE1705>

Ok it's what I guessed.

Thanks
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Tue Mar  9 08:51:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 10:51:30 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B966C8E.9040805@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
	<4B955401.1030004@marketelsystems.com>
	<4B966C8E.9040805@marketelsystems.com>
Message-ID: <191c3a031003090851m66f89d6ey9e20cb8d8f064852@mail.gmail.com>

you are missing:

call-command: execute

and optionally:

event-lock: true // if you want this event to be exclusive until its done
and
async: true // if you are in sync mode and you want to do the command async


you might want to look at the ESL lib and use that instead of rolling your
own event socket code which
usually generates many question and answer sessions like this one.




-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From infos at madovsky.org  Tue Mar  9 08:55:29 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 11:55:29 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<F996975CC7C34E28A54D55CEBC5E4CB0@MOBILEE1705>
Message-ID: <EE8802DA199140B581FACD5359022E9A@MOBILEE1705>

I Have no money, but a nice Gibson Les Paul standard from 1990 to offer if
my experimental project will end soon...
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:47 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Ok it's what I guessed.

  Thanks
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 09, 2010 11:39 AM
    Subject: Re: [Freeswitch-users] vars from sip_registrations table


    You would need to make a patch for it or pay someone to add it.



    On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

      forgot to say that I need it (since I didn't find any good examples)
      for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

      Thanks

      F
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 2:59 AM
        Subject: vars from sip_registrations table


        in sip_registration table there are
              call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

        fields, so how can I get these vars from the callee before the bridge ?
        I need to replace the host of the callee in case of if he's registered on another FS node.

        thanks

        Franck

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





    -- 
    Anthony Minessale II

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

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

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



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


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From dave.redmore at spigotsystems.com  Tue Mar  9 08:44:03 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Tue, 9 Mar 2010 10:44:03 -0600 (CST)
Subject: [Freeswitch-users] Freeswitch a good fit?
In-Reply-To: <22405606.601268153039310.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>

Hi All, 

New to the list and just starting to look at Freeswitch. 

I have an application that I'm looking at and wondering if Freeswitch is the right fit. 

I need a (fairly) simple SIP server and a T.38 media proxy. The setup would be SIP FXS units at the customers location doing outbound faxing. via T.38. The FXS needs to register with my server, which will authenticate the user and connect the call to my T.38 provider's gateway. 

I've seen some reference to using a "proxy-media" setting in the WIKI, but refers to it being a "hack" - so not a big confidence builder. 

I've also looked at using OpenSIPS and MediaProxy, which looks to do what I'm trying to accomplish as well. 

Does it make sense to deploy FS in this scenario? 

Thanks, 

Dave Redmore 
Spigot Networks, Inc. 
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From infos at madovsky.org  Tue Mar  9 09:37:03 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:37:03 -0500
Subject: [Freeswitch-users] Eavesdrop
References: <4B95244A.8030808@marketelsystems.com><b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com><4B955401.1030004@marketelsystems.com><4B966C8E.9040805@marketelsystems.com>
	<191c3a031003090851m66f89d6ey9e20cb8d8f064852@mail.gmail.com>
Message-ID: <BD0D7EF57A5142B7B28B7D430F5A1DEA@MOBILEE1705>

also I would love to make a patch, but i"m really not C or C++ guru...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:51 AM
  Subject: Re: [Freeswitch-users] Eavesdrop


  you are missing:

  call-command: execute

  and optionally:

  event-lock: true // if you want this event to be exclusive until its done
  and
  async: true // if you are in sync mode and you want to do the command async


  you might want to look at the ESL lib and use that instead of rolling your own event socket code which
  usually generates many question and answer sessions like this one.




  -- 
  Anthony Minessale II

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

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

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



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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Mailings at kh-dev.de  Tue Mar  9 09:36:54 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 18:36:54 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
	<191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4D@srv02.khdev.corp>

Good idea...

But why don't you send a fax to everyone...   ;-)


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Tuesday, March 09, 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

All these threads seem to go off on on tangents, seems at least a lot of people are reading it so.....

This thread brought to you by ClueCon MMX!
register today http://www.cluecon.com

Hey, it was worth a shot, maybe we can do a presentation on t38 in FS (if we ever get the time to finish it).



On Mon, Mar 8, 2010 at 5:30 PM, Tod Hansmann <freeswitch.org<http://freeswitch.org>@todandlorna.com<http://todandlorna.com>> wrote:
There exists a large number of companies that make large profits largely because they sell based on what they think the customer will pay.  NDAs keep customers from comparing prices, and competitors from competing on price.  This has been a pretty common telecom practice, though not NDA specifically.  This is why the "let me see a copy of your phone bill" line is so ill-advised to be substantiated when talking to a new provider.  If you give them your phone bill, they know exactly where the bar must be set to win.

-Tod Hansmann


On 3/8/2010 4:05 PM, Brian West wrote:
Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me...

/b

On Mar 8, 2010, at 5:01 PM, David Knell wrote:


And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

--Dave





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

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
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From infos at madovsky.org  Tue Mar  9 09:37:49 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:37:49 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <DABE813E4C5D4AA58E163ECB00355A24@MOBILEE1705>

soooorry I answered to the wrong thread in my last email
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

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



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


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From infos at madovsky.org  Tue Mar  9 09:39:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:39:18 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>

OK How much ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

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



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


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From anthony.minessale at gmail.com  Tue Mar  9 09:47:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 11:47:24 -0600
Subject: [Freeswitch-users] Freeswitch a good fit?
In-Reply-To: <6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>
References: <22405606.601268153039310.JavaMail.root@zimbra1.spigotsystems.com>
	<6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <191c3a031003090947r384d5b05ve9e534df1db8ea2@mail.gmail.com>

Its not a hack in the sense that it's not likely to work.  It's a hack in
the sense that FS acts mostly as a b2bua role and proxy is a feature mostly
left to proxy software.

I would setup both and compare, using FS might be overkill if you are not
planning to ever do anything else besides proxy t38.



On Tue, Mar 9, 2010 at 10:44 AM, Dave Redmore <
dave.redmore at spigotsystems.com> wrote:

> Hi All,
>
> New to the list and just starting to look at Freeswitch.
>
> I have an application that I'm looking at and wondering if Freeswitch is
> the right fit.
>
> I need a (fairly) simple SIP server and a T.38 media proxy.  The setup
> would be SIP FXS units at the customers location doing outbound faxing. via
> T.38.  The FXS needs to register with my server, which will authenticate the
> user and connect the call to my T.38 provider's gateway.
>
> I've seen some reference to using a "proxy-media" setting in the WIKI, but
> refers to it being a "hack" - so not a big confidence builder.
>
> I've also looked at using OpenSIPS and MediaProxy, which looks to do what
> I'm trying to accomplish as well.
>
> Does it make sense to deploy FS in this scenario?
>
> Thanks,
>
> Dave Redmore
> Spigot Networks, Inc.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From sekharreddypk at gmail.com  Tue Mar  9 09:40:37 2010
From: sekharreddypk at gmail.com (chandrasekhar reddy)
Date: Tue, 9 Mar 2010 23:10:37 +0530
Subject: [Freeswitch-users] call forward
Message-ID: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>

Hi,

i am trying to forward call from freeswitch to cisco router, call is
estabilished but after 30sec call got ended, any idea

sekhar
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From sean at obscuradigital.com  Tue Mar  9 09:59:18 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 09 Mar 2010 09:59:18 -0800
Subject: [Freeswitch-users] Outbound calls mobile
Message-ID: <C7BBCC76.5FEF%sean@obscuradigital.com>

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean
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From andrew at hijacked.us  Tue Mar  9 10:06:00 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Mar 2010 13:06:00 -0500
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBCC76.5FEF%sean@obscuradigital.com>
References: <C7BBCC76.5FEF%sean@obscuradigital.com>
Message-ID: <20100309180600.GG18802@hijacked.us>

On Tue, Mar 09, 2010 at 09:59:18AM -0800, Sean Holt wrote:
> Hello, having weird issue with outbound calls to any mobile phone, but not
> to LAN/analog lines.  Curious if anyone else has had this experienced with
> their freeswitch setup.
> 
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
>
Make sure the CallerID you're sending isn't making the provider reject
it. I've seen this happen several times.

Andrew


From peder at networkoblivion.com  Tue Mar  9 10:08:35 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Mar 2010 12:08:35 -0600
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBCC76.5FEF%sean@obscuradigital.com>
References: <C7BBCC76.5FEF%sean@obscuradigital.com>
Message-ID: <107001cabfb3$891cd5e0$9b5681a0$@com>

What are the problems?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 11:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Outbound calls mobile

 

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean 

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From anthony.minessale at gmail.com  Tue Mar  9 10:10:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 12:10:42 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
Message-ID: <191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>

Aww, The les paul sounded nice I have one on my wishlist =D
http://bit.ly/bRHKlk

*shrug* $250.00 maybe?


On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

>  OK How much ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 11:39 AM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> You would need to make a patch for it or pay someone to add it.
>
>
> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  forgot to say that I need it (since I didn't find any good examples)
>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>> not compatible)
>>
>> Thanks
>>
>> F
>>
>>  ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>> *Subject:* vars from sip_registrations table
>>
>> in sip_registration table there are
>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>
>> fields, so how can I get these vars from the callee before the bridge ?
>> I need to replace the host of the callee in case of if he's registered on
>> another FS node.
>>
>> thanks
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  9 10:25:40 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 13:25:40 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
Message-ID: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>

I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
I understand perfectly your mind of how to manage an open source project, but don't forget
that there are guys like you on this emailist who don't even make any decent salary, that is why
I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
but not                                         share -> ask money -> make people weird -> money won't come
it's my point of view.

Regards
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:10 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Aww, The les paul sounded nice I have one on my wishlist =D 
  http://bit.ly/bRHKlk

  *shrug* $250.00 maybe?



  On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

    OK How much ?
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 11:39 AM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      You would need to make a patch for it or pay someone to add it.



      On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

        forgot to say that I need it (since I didn't find any good examples)
        for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

        Thanks

        F
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 2:59 AM
          Subject: vars from sip_registrations table


          in sip_registration table there are
                call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

          fields, so how can I get these vars from the callee before the bridge ?
          I need to replace the host of the callee in case of if he's registered on another FS node.

          thanks

          Franck

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





      -- 
      Anthony Minessale II

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

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

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



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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sean at obscuradigital.com  Tue Mar  9 10:30:26 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 09 Mar 2010 10:30:26 -0800
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <107001cabfb3$891cd5e0$9b5681a0$@com>
Message-ID: <C7BBD3C2.5FF6%sean@obscuradigital.com>


Basically I have sip trace turned on and I?m watching the call.  It gets to
the point where it is ?Trying? to esbstalish the call then for no reason it
times out.  No further details from the sip trace other then call has timed
out.

The strange part is this doesn?t happen when calling any analog line.

If you want I can provide the sip trace.

Sean


On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:

> What are the problems?
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
> Sent: Tuesday, March 09, 2010 11:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Outbound calls mobile
>  
> Hello, having weird issue with outbound calls to any mobile phone, but not to
> LAN/analog lines.  Curious if anyone else has had this experienced with their
> freeswitch setup.
> 
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
> 
> Thanks
> Sean 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Mar  9 10:45:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 12:45:02 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
Message-ID: <191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>

ORLY,

So maybe if you *think* you know what you are talking about, you should go
do some research on the average amount of code I write on a daily basis
nonstop for the last several years without "ask money".  This includes most
of the core, and 90% of the modules you are playing with right now including
the 24 thousand lines of code in mod_sofia alone.
So frankly, since I don't see an urgent need for your request, and I am too
busy to do it, this particular feature would require a bounty to complete.

By the way, you have sent like 2 dozen emails in the same week asking for
all kinds of help, that's also being shared to you for free so I think you
are getting the better side of the bargain here.  Asking for a bounty on a
feature that only helps one individual is very common.............




On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

>  I said I OFFER it in case of I finish my (no money no profit but big
> potential) experimental project.
> I understand perfectly your mind of how to manage an open source project,
> but don't forget
> that there are guys like you on this emailist who don't even make any
> decent salary, that is why
> I use and develop open source : share -> evovle your minds -> make all
> people happy -> money comes
> but not                                         share -> ask money -> make
> people weird -> money won't come
> it's my point of view.
>
> Regards
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 1:10 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Aww, The les paul sounded nice I have one on my wishlist =D
> http://bit.ly/bRHKlk
>
> *shrug* $250.00 maybe?
>
>
> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  OK How much ?
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>  *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> You would need to make a patch for it or pay someone to add it.
>>
>>
>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  forgot to say that I need it (since I didn't find any good examples)
>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>>> not compatible)
>>>
>>> Thanks
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>> *Subject:* vars from sip_registrations table
>>>
>>> in sip_registration table there are
>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>
>>> fields, so how can I get these vars from the callee before the bridge ?
>>> I need to replace the host of the callee in case of if he's registered on
>>> another FS node.
>>>
>>> thanks
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From peder at networkoblivion.com  Tue Mar  9 10:47:32 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Mar 2010 12:47:32 -0600
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBD3C2.5FF6%sean@obscuradigital.com>
References: <107001cabfb3$891cd5e0$9b5681a0$@com>
	<C7BBD3C2.5FF6%sean@obscuradigital.com>
Message-ID: <10e701cabfb8$fa211440$ee633cc0$@com>

Is this out a sip trunk, or a PRI, or some other gateway?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 12:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound calls mobile

 


Basically I have sip trace turned on and I'm watching the call.  It gets to
the point where it is "Trying" to esbstalish the call then for no reason it
times out.  No further details from the sip trace other then call has timed
out.

The strange part is this doesn't happen when calling any analog line.

If you want I can provide the sip trace.

Sean


On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:

What are the problems?
 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 11:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Outbound calls mobile

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean 

  _____  

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

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From msc at freeswitch.org  Tue Mar  9 11:01:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 11:01:31 -0800
Subject: [Freeswitch-users] auto attendant -> hunt group -> external
	number does not ring
In-Reply-To: <302537.41702.qm@web57609.mail.re1.yahoo.com>
References: <302537.41702.qm@web57609.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:23 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> FS Community,
>
>
> I created an extension 704
>
>
>
> I created a hunt group 7004 that first rings 704 then rings my cell phone
> 16194548924.
>
>
>
> I created an auto attendant 5002 with an option of 704 that goes to hunt
> group 7004.
>
>
>
> 1)     What does work: When I call the hunt group directly by dialing 7004
> internally, the hunt group first rings 704 then rings my cell phone
> successfully.  This is perfect.
>
>
>
> 2)     What does not work: When I call the main auto attendant from either
> internally by dialing 5002 or externally by dialing my main number, and dial
> 704, my extension rings and after not answering my cell phone does not ring.
>
>
>
> What I am expecting in #2 is for my extension 704 to ring and if I do not
> pick up then my cell phone will ring just like it does in #1.
>
>
>
> My knowledge is limited in this area and if you could point me in the right
> direction, that would be great.  I a have tried everything but with no
> luck.
>
>
> Pastebin the necessary debug info. Check
http://wiki.freeswitch.org/wiki/Reporting_Bugs for specifics on how to do
collect and report all the data to pastebin. It would be good to see the
dialplan and the debug log for each call.

When you've collected all the data please reply here with the pastebin URL.
-MC
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From anthony.minessale at gmail.com  Tue Mar  9 11:06:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 13:06:07 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>

Also, when you are done insulting me, maybe try sofia_contact api command
that will resolve your destination to the exact contact string that the
phone registered with which is really the only bit of info
that you would possibly need rather than asking for another expensive
database lookup per call.


On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ORLY,
>
> So maybe if you *think* you know what you are talking about, you should go
> do some research on the average amount of code I write on a daily basis
> nonstop for the last several years without "ask money".  This includes most
> of the core, and 90% of the modules you are playing with right now including
> the 24 thousand lines of code in mod_sofia alone.
> So frankly, since I don't see an urgent need for your request, and I am too
> busy to do it, this particular feature would require a bounty to complete.
>
> By the way, you have sent like 2 dozen emails in the same week asking for
> all kinds of help, that's also being shared to you for free so I think you
> are getting the better side of the bargain here.  Asking for a bounty on a
> feature that only helps one individual is very common.............
>
>
>
>
>
> On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I said I OFFER it in case of I finish my (no money no profit but big
>> potential) experimental project.
>> I understand perfectly your mind of how to manage an open source project,
>> but don't forget
>> that there are guys like you on this emailist who don't even make any
>> decent salary, that is why
>> I use and develop open source : share -> evovle your minds -> make all
>> people happy -> money comes
>> but not                                         share -> ask money -> make
>> people weird -> money won't come
>> it's my point of view.
>>
>> Regards
>>
>> ----- Original Message -----
>>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 1:10 PM
>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> Aww, The les paul sounded nice I have one on my wishlist =D
>> http://bit.ly/bRHKlk
>>
>> *shrug* $250.00 maybe?
>>
>>
>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  OK How much ?
>>>
>>>  ----- Original Message -----
>>>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> You would need to make a patch for it or pay someone to add it.
>>>
>>>
>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>> is not compatible)
>>>>
>>>> Thanks
>>>>
>>>> F
>>>>
>>>>  ----- Original Message -----
>>>>  *From:* Madovsky <infos at madovsky.org>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>> *Subject:* vars from sip_registrations table
>>>>
>>>> in sip_registration table there are
>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>
>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>> I need to replace the host of the callee in case of if he's registered
>>>> on another FS node.
>>>>
>>>> thanks
>>>>
>>>> Franck
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rm at callrica.co.za  Tue Mar  9 11:08:42 2010
From: rm at callrica.co.za (Roly Maz)
Date: Tue, 9 Mar 2010 21:08:42 +0200
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
Message-ID: <000001cabfbb$f2114c90$d633e5b0$@co.za>

Ah I see... what? Please share and lead this blind man out the FS
wilderness!

I don't understand...what happens to the external profile? Do you delete it?
And how do you forward port 5060? 

...and you thought you were a newbie!

Any insight would be much appreciated...loving the journey.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 09 March 2010 08:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

Ah I see.  I will try again using the internal profile and forwarding  
port 5060.  Presumably still creating a directory entry to enable the  
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two  
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing  
>> something wrong here as I am still having difficulty, and still  
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding  
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to  
>> retain an entry in the directory /usr/local/freeeswitch/conf/ 
>> directory/ext_dns.xml in order to give freeswitch the dns name of  
>> the server as a domain for the remote softphone to register on.  I  
>> left the group name in this entry the same as inthe default entry,  
>> so that the remote phone could register on the same extension  
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN  
>> as freeswitch) phone to the remote phone, I get the message on the  
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of  
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment  
>> <param name="aggressive-nat-detection" value="true"/> in  
>> external.xml, otherwise in the case of a call initiated by the  
>> remote phone being hung up by the local phone, freeswitch sent the  
>> BYE to the private IP of the remote phone, rather than its public  
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Mar  9 11:09:06 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:09:06 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <AF70391FDBF24706A96129C21BF4C3F1@MOBILEE1705>

>This includes most of the core, and 90% of the modules you are playing with ..
do you think I really "play" with ? it's like you are a teacher, complaints about your knowledge and complaint to your students
you worked for nothing for years. 
24000 lines ? it's nothing if I compare to a project I offer fro free for NGOs  that contains 800 000.lines do you want to continue to compete
and complaint ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:45 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  ORLY,

  So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
  So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

  By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............





  On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

    I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
    I understand perfectly your mind of how to manage an open source project, but don't forget
    that there are guys like you on this emailist who don't even make any decent salary, that is why
    I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
    but not                                         share -> ask money -> make people weird -> money won't come
    it's my point of view.

    Regards
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 1:10 PM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      Aww, The les paul sounded nice I have one on my wishlist =D 
      http://bit.ly/bRHKlk

      *shrug* $250.00 maybe?



      On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

        OK How much ?
          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 11:39 AM
          Subject: Re: [Freeswitch-users] vars from sip_registrations table


          You would need to make a patch for it or pay someone to add it.



          On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

            forgot to say that I need it (since I didn't find any good examples)
            for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 09, 2010 2:59 AM
              Subject: vars from sip_registrations table


              in sip_registration table there are
                    call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

              fields, so how can I get these vars from the callee before the bridge ?
              I need to replace the host of the callee in case of if he's registered on another FS node.

              thanks

              Franck

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          -- 
          Anthony Minessale II

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

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

          FreeSWITCH Developer Conference
          sip:888 at conference.freeswitch.org
          iax:guest at conference.freeswitch.org/888
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          pstn:+19193869900



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      -- 
      Anthony Minessale II

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

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

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



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


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  -- 
  Anthony Minessale II

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

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
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From sean at obscuradigital.com  Tue Mar  9 11:08:38 2010
From: sean at obscuradigital.com (Obscura)
Date: Tue, 9 Mar 2010 11:08:38 -0800
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <10e701cabfb8$fa211440$ee633cc0$@com>
References: <107001cabfb3$891cd5e0$9b5681a0$@com>
	<C7BBD3C2.5FF6%sean@obscuradigital.com>
	<10e701cabfb8$fa211440$ee633cc0$@com>
Message-ID: <90B41691-BD5F-4E36-B8D6-B394889E3ADF@obscuradigital.com>

I'm using iCall carrier service gateway.  IP based authentication.  I  
know it's worked in the past but ever since I did a svn update it  
hasn't worked speifically connecting to mobile/cell phones.  Not sure  
if this answers question.

Sean
Sent from my iPhone
On Mar 9, 2010, at 10:47 AM, "Peder" <peder at networkoblivion.com> wrote:

> Is this out a sip trunk, or a PRI, or some other gateway?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Sean Holt
> Sent: Tuesday, March 09, 2010 12:30 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound calls mobile
>
>
>
>
> Basically I have sip trace turned on and I?m watching the call.  It  
> gets to the point where it is ?Trying? to esbstalish the call  
> then for no reason it times out.  No further details from the sip tr 
> ace other then call has timed out.
>
> The strange part is this doesn?t happen when calling any analog lin 
> e.
>
> If you want I can provide the sip trace.
>
> Sean
>
>
> On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:
>
> What are the problems?
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Sean Holt
> Sent: Tuesday, March 09, 2010 11:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Outbound calls mobile
>
> Hello, having weird issue with outbound calls to any mobile phone,  
> but not to LAN/analog lines.  Curious if anyone else has had this  
> experienced with their freeswitch setup.
>
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
>
> Thanks
> Sean
>
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From infos at madovsky.org  Tue Mar  9 11:11:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:11:58 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <3C08E64A35614EE68E5936C80E809F9B@MOBILEE1705>

In more let me say to you that my "dozen" of emails  I sent to this emailist were not "help"
but info request because of doc lacks... So anyway, I don't care what do you think about me.
but the thing I'm sure is I don't complaint of my situation since I'm not the center of the earth...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:45 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  ORLY,

  So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
  So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

  By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............





  On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

    I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
    I understand perfectly your mind of how to manage an open source project, but don't forget
    that there are guys like you on this emailist who don't even make any decent salary, that is why
    I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
    but not                                         share -> ask money -> make people weird -> money won't come
    it's my point of view.

    Regards
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 1:10 PM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      Aww, The les paul sounded nice I have one on my wishlist =D 
      http://bit.ly/bRHKlk

      *shrug* $250.00 maybe?



      On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

        OK How much ?
          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 11:39 AM
          Subject: Re: [Freeswitch-users] vars from sip_registrations table


          You would need to make a patch for it or pay someone to add it.



          On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

            forgot to say that I need it (since I didn't find any good examples)
            for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 09, 2010 2:59 AM
              Subject: vars from sip_registrations table


              in sip_registration table there are
                    call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

              fields, so how can I get these vars from the callee before the bridge ?
              I need to replace the host of the callee in case of if he's registered on another FS node.

              thanks

              Franck

            _______________________________________________
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          -- 
          Anthony Minessale II

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

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

          FreeSWITCH Developer Conference
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          iax:guest at conference.freeswitch.org/888
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      -- 
      Anthony Minessale II

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

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

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

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
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From infos at madovsky.org  Tue Mar  9 11:14:53 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:14:53 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705><191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
Message-ID: <B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>

Ok, tell me where I insulted you ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:06 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Also, when you are done insulting me, maybe try sofia_contact api command that will resolve your destination to the exact contact string that the phone registered with which is really the only bit of info
  that you would possibly need rather than asking for another expensive database lookup per call.



  On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

    ORLY,

    So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
    So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

    By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............






    On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

      I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
      I understand perfectly your mind of how to manage an open source project, but don't forget
      that there are guys like you on this emailist who don't even make any decent salary, that is why
      I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
      but not                                         share -> ask money -> make people weird -> money won't come
      it's my point of view.

      Regards
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 1:10 PM
        Subject: Re: [Freeswitch-users] vars from sip_registrations table


        Aww, The les paul sounded nice I have one on my wishlist =D 
        http://bit.ly/bRHKlk

        *shrug* $250.00 maybe?



        On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

          OK How much ?
            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 09, 2010 11:39 AM
            Subject: Re: [Freeswitch-users] vars from sip_registrations table


            You would need to make a patch for it or pay someone to add it.



            On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

              forgot to say that I need it (since I didn't find any good examples)
              for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

              Thanks

              F
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 09, 2010 2:59 AM
                Subject: vars from sip_registrations table


                in sip_registration table there are
                      call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

                fields, so how can I get these vars from the callee before the bridge ?
                I need to replace the host of the callee in case of if he's registered on another FS node.

                thanks

                Franck

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            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
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            AIM: anthm
            MSN:anthony_minessale at hotmail.com
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        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
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        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
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    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
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  -- 
  Anthony Minessale II

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

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From infos at madovsky.org  Tue Mar  9 11:16:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:16:08 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705><191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
Message-ID: <49F280E71F554CFD8BDCD78E53F14A71@MOBILEE1705>

Thank you !
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:06 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Also, when you are done insulting me, maybe try sofia_contact api command that will resolve your destination to the exact contact string that the phone registered with which is really the only bit of info
  that you would possibly need rather than asking for another expensive database lookup per call.



  On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

    ORLY,

    So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
    So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

    By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............






    On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

      I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
      I understand perfectly your mind of how to manage an open source project, but don't forget
      that there are guys like you on this emailist who don't even make any decent salary, that is why
      I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
      but not                                         share -> ask money -> make people weird -> money won't come
      it's my point of view.

      Regards
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 1:10 PM
        Subject: Re: [Freeswitch-users] vars from sip_registrations table


        Aww, The les paul sounded nice I have one on my wishlist =D 
        http://bit.ly/bRHKlk

        *shrug* $250.00 maybe?



        On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

          OK How much ?
            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 09, 2010 11:39 AM
            Subject: Re: [Freeswitch-users] vars from sip_registrations table


            You would need to make a patch for it or pay someone to add it.



            On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

              forgot to say that I need it (since I didn't find any good examples)
              for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

              Thanks

              F
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 09, 2010 2:59 AM
                Subject: vars from sip_registrations table


                in sip_registration table there are
                      call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

                fields, so how can I get these vars from the callee before the bridge ?
                I need to replace the host of the callee in case of if he's registered on another FS node.

                thanks

                Franck

              _______________________________________________
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              UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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            -- 
            Anthony Minessale II

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

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
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        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

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

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

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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    FreeSWITCH Developer Conference
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    iax:guest at conference.freeswitch.org/888
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  -- 
  Anthony Minessale II

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

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

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
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From msc at freeswitch.org  Tue Mar  9 11:18:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 11:18:39 -0800
Subject: [Freeswitch-users] call forward
In-Reply-To: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>
References: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>
Message-ID: <87f2f3b91003091118o1191c519wb2972c1c8f9fc087@mail.gmail.com>

On Tue, Mar 9, 2010 at 9:40 AM, chandrasekhar reddy <sekharreddypk at gmail.com
> wrote:

> Hi,
>
> i am trying to forward call from freeswitch to cisco router, call is
> estabilished but after 30sec call got ended, any idea
>
>

Sounds like a timeout issue. Make sure that there's no naughty NAT router in
between your devices. You could also get a pcap of the traffic between the
two devices and see where the dialog goes awry.
-MC
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From anthony.minessale at gmail.com  Tue Mar  9 11:41:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 13:41:30 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
	<B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>
Message-ID: <191c3a031003091141i32599500o338f86d225bd3cb9@mail.gmail.com>

Well every email you sent since i suggested you file a bounty request for
your feature is more or less carrying an insulting tone.  Go back and read
it.  You are insulting everyone on this list with your attempts to downplay
the effort the community has made to create this project.

An on the subject of being able to read, I said "mod_sofia" (1 module in FS)
is 24000 lines.
This is not a contest about how many lines of code but since you feel the
need to continue:

http://fisheye.freeswitch.org/committer/FreeSWITCH/anthm

FreeSWITCH has over 3 million lines of code in SVN.

You can clearly see I have personally committed 600,000 lines of code in 4
years and started this project from nothing.
The only point of this information is that almost all of this code was
written completely for free.
This means you are completely wrong with your previous statements that
everything is for money.
The actual amount of code written and time spent is not even relevant apart
from that.

I only ask for money for things I don't want to do and someone else
wants me to do it.  In fact, even if you paid me, I would not add your
feature so forget it.
The more I thought about it the more I decided it's a waste of resources on
the application.

If our work is insignificant to you, I don't really care.  Feel free to
write your own code instead.
Keep in mind that if you continue to abuse and flame our mailing list, you
will be moderated.

If you can be civil and stop the flaming you are welcome to stay.

I recommend an apology to the community to start with.
Also, try to consolidate your thoughts into 1 single email instead of
sending 5 in a row on the same thread.


On Tue, Mar 9, 2010 at 1:14 PM, Madovsky <infos at madovsky.org> wrote:

>  Ok, tell me where I insulted you ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 2:06 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Also, when you are done insulting me, maybe try sofia_contact api command
> that will resolve your destination to the exact contact string that the
> phone registered with which is really the only bit of info
> that you would possibly need rather than asking for another expensive
> database lookup per call.
>
>
> On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> ORLY,
>>
>> So maybe if you *think* you know what you are talking about, you should go
>> do some research on the average amount of code I write on a daily basis
>> nonstop for the last several years without "ask money".  This includes most
>> of the core, and 90% of the modules you are playing with right now including
>> the 24 thousand lines of code in mod_sofia alone.
>> So frankly, since I don't see an urgent need for your request, and I am
>> too busy to do it, this particular feature would require a bounty to
>> complete.
>>
>> By the way, you have sent like 2 dozen emails in the same week asking for
>> all kinds of help, that's also being shared to you for free so I think you
>> are getting the better side of the bargain here.  Asking for a bounty on a
>> feature that only helps one individual is very common.............
>>
>>
>>
>>
>>
>> On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  I said I OFFER it in case of I finish my (no money no profit but big
>>> potential) experimental project.
>>> I understand perfectly your mind of how to manage an open source project,
>>> but don't forget
>>> that there are guys like you on this emailist who don't even make any
>>> decent salary, that is why
>>> I use and develop open source : share -> evovle your minds -> make all
>>> people happy -> money comes
>>> but not                                         share -> ask money ->
>>> make people weird -> money won't come
>>> it's my point of view.
>>>
>>> Regards
>>>
>>>  ----- Original Message -----
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 1:10 PM
>>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> Aww, The les paul sounded nice I have one on my wishlist =D
>>> http://bit.ly/bRHKlk
>>>
>>> *shrug* $250.00 maybe?
>>>
>>>
>>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  OK How much ?
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>>
>>>> You would need to make a patch for it or pay someone to add it.
>>>>
>>>>
>>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>>> is not compatible)
>>>>>
>>>>> Thanks
>>>>>
>>>>> F
>>>>>
>>>>>  ----- Original Message -----
>>>>> *From:* Madovsky <infos at madovsky.org>
>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>>> *Subject:* vars from sip_registrations table
>>>>>
>>>>> in sip_registration table there are
>>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>>
>>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>>> I need to replace the host of the callee in case of if he's registered
>>>>> on another FS node.
>>>>>
>>>>> thanks
>>>>>
>>>>> Franck
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> ------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Tue Mar  9 12:43:37 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 9 Mar 2010 21:43:37 +0100
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
Message-ID: <4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>

This is hypochisy  in action.

99% of people asking questions in this ml are using FreeSWITCH for
commercial use, I for one.
So if the primary author of software that gets you paid asks for small
bounty, I'd shut up or pay up.


Ognjen

On Tue, Mar 9, 2010 at 7:25 PM, Madovsky <infos at madovsky.org> wrote:

>  I said I OFFER it in case of I finish my (no money no profit but big
> potential) experimental project.
> I understand perfectly your mind of how to manage an open source project,
> but don't forget
> that there are guys like you on this emailist who don't even make any
> decent salary, that is why
> I use and develop open source : share -> evovle your minds -> make all
> people happy -> money comes
> but not                                         share -> ask money -> make
> people weird -> money won't come
> it's my point of view.
>
> Regards
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 1:10 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Aww, The les paul sounded nice I have one on my wishlist =D
> http://bit.ly/bRHKlk
>
> *shrug* $250.00 maybe?
>
>
> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  OK How much ?
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>  *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> You would need to make a patch for it or pay someone to add it.
>>
>>
>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  forgot to say that I need it (since I didn't find any good examples)
>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>>> not compatible)
>>>
>>> Thanks
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>> *Subject:* vars from sip_registrations table
>>>
>>> in sip_registration table there are
>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>
>>> fields, so how can I get these vars from the callee before the bridge ?
>>> I need to replace the host of the callee in case of if he's registered on
>>> another FS node.
>>>
>>> thanks
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Tue Mar  9 13:57:01 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 15:57:01 -0600
Subject: [Freeswitch-users] IVR and Erlang
In-Reply-To: <4B703921.4080909@marketelsystems.com>
References: <4B70288F.6040003@marketelsystems.com>
	<4B703921.4080909@marketelsystems.com>
Message-ID: <4B96C42D.5020609@marketelsystems.com>

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From m.sobkow at marketelsystems.com  Tue Mar  9 14:46:16 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 16:46:16 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304234050.GL1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>	<20100304184755.GK1751@hijacked.us>	<4B9035B4.6030901@marketelsystems.com>	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<20100304234050.GL1751@hijacked.us>
Message-ID: <4B96CFB8.6060803@marketelsystems.com>

I spoke with "the boss" today, and he's hoping to do a donation at month 
end if our accountant can confirm that it can be taken as a tax writeoff 
(I suggested it's an alternative way of paying for software, so it 
should be write-offable.)

Andrew Thompson wrote:
> On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:
>
>   
>>> "Another satisfied customer!" :D Now it's time to throw some of that
>>>       
>> Canadian colored money at Tony's paypal account. ;)
>>     
>
> Hey! I wrote the erlang module and provided the solution to the problem
> ;)
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


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



From craig at overthewire.com.au  Tue Mar  9 15:25:31 2010
From: craig at overthewire.com.au (Craig Askings)
Date: Wed, 10 Mar 2010 09:25:31 +1000
Subject: [Freeswitch-users] freeswitch to cisco router
In-Reply-To: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
References: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
Message-ID: <8cc991dd1003091525t2600adf1m3c45dbf5ef62e7e6@mail.gmail.com>

Hi Swapna,

Are you trying to register the Cisco to the Freeswitch server? or the other
way around. If the former check out the following documentation.

http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter1.html#wp1182326

sip-ua
  authentication username [INSERT_EXTENTION] password [INSERT_PASSWORD]
  registrar ipv4:[INERST_NAME_OF_FS_BOX]:5060 expires 3600
  sip-server ipv4:[INERST_NAME_OF_FS_BOX]:5060

Regards,
<http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter1.html#wp1182326>
Craig.

On 9 March 2010 23:47, srinivasula reddy <srinivas.ksvreddy at gmail.com>wrote:

> Hi,
>
> i have tried to configure gateway from my freeswitch, with the following
> credentials proxy(ip of cisco) and username,password.
> and when i start my freeswitch registration reqest is going to cisco router
> and again it requesting to same registration reqeust to my sipserver, the
> register packet from cisco router to my freeswitch is comming like this
>
> To: sip:.*@sipserverip:5060
>
> so there there is no user present with the name of .* it is rejecting the
> registration request, any idea
>
> Thanks
> Swapna
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From lloyd.aloysius at gmail.com  Tue Mar  9 16:57:33 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 9 Mar 2010 19:57:33 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
Message-ID: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>

Hi All,

I am trying to setup a simple IVR with two options , timeout action and
Invalid digits pressed. here is the partial IVR Menu.


<entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
default"/>
<entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
default"/>


How to Catch Timeout and Invalid Digits Actions from the IVR menu?

Thanks you

Thanks,
Lloyd
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From frank at carmickle.com  Tue Mar  9 17:17:40 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 9 Mar 2010 20:17:40 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
In-Reply-To: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
References: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
Message-ID: <20100310011738.GS18427@base.carmickle.com>

Hello

On Tue, Mar 09, Aloysius Lloyd wrote:
> Hi All,
> 
> I am trying to setup a simple IVR with two options , timeout action and
> Invalid digits pressed. here is the partial IVR Menu.
> 
> 
> <entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
> default"/>
> <entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
> default"/>

Add something like.

                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
 
Then in the dialplan after you've called the ivr tranfer to another exten.  Be careful to make sure that hangup after bridge gets set so that you don't ring the exten when someone who has been called from the ivr hangs up.

HTH
--FC


From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:20:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:20:02 -0500
Subject: [Freeswitch-users] How to tie context to a gateway?
In-Reply-To: <20100216141634.75B3511FC6@mail.nstel.ru>
References: <20100216141634.75B3511FC6@mail.nstel.ru>
Message-ID: <7BA30ABD-5886-42AF-B22F-FBC4D13FDCE0@jerris.com>

You can use conditions and transfer, the issue is, the inbound call has to be matched on the gateway, which means the other endpoint must actually send the request to our registered contact.  Many do not do this.  Its probably easier to just use multiple profiles on multiple ports.

Mike


On Feb 16, 2010, at 9:16 AM, Nikolay Kondratyev wrote:

> Hi all,
> I have several gateways in the external profile.
> Let?s say GW1 and GW2. I?d like to process calls from the GW1 in the context C1 and calls from GW2 in the context C2.
> Parameter ?context?, as far as I understand works for the whole profile, not for individual gateways in the profile.
> How do send calls from GW1 into context C1?
> What will be a good practice to do that?
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Mar  9 16:21:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:21:32 -0500
Subject: [Freeswitch-users] call routing from freeswitch based on INVITE
In-Reply-To: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>
References: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>
Message-ID: <EEB17718-A279-4E5F-BBF1-5C60765A43A3@jerris.com>

if you look at the vars from the info app for that inbound call, you will see all the relevant parts of the sip packet there to use in a condition.

Mike

On Mar 1, 2010, at 2:39 AM, srinivasula reddy wrote:

>  
>  
> I have two sipservers like server1 and server2,  if sever1 receives invite packet like
>  
> INVITE From:   1000 at server1.domain.com
>            To:   1002 at server2.railvoice.com.
>  
> how can i route the invite packet to server2 from server1, 

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From mike at jerris.com  Tue Mar  9 16:25:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:25:55 -0500
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
	<alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
Message-ID: <034FE08F-7E01-45DD-8758-61FEA5AEEAF1@jerris.com>

Could you please contribute a patch or at least report this issue to http;//jira.frreeswitch,org

On Mar 1, 2010, at 10:29 PM, Wellie Chao wrote:

> Terrific. "show calls as xml" works great and just what I needed.
> 
> P.S. "show calls as csv" still has the comma bug. Not a big deal since I 
> can get it as XML. CSV needs quotes around field values that contain a 
> comma. Pretty easy to fix. Few lines of C (strchr for "," and if found, 
> put quotes around field value).
> 



From pjintheusa at gmail.com  Tue Mar  9 18:15:15 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Mar 2010 21:15:15 -0500
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
	<9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>
Message-ID: <367751821003091815q72cce2bey6399b90c657a61f8@mail.gmail.com>

After originate I call a method:

Uuid = fsApi.ExecuteString("create_uuid");
apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/g1/{2}
'&managed(MyFSProjects.TestCallAnswered)'", Uuid, OutgoingCallerID,
NumberToDial));

My "TestCallAnswered" has reference to the session:

public void Run(AppContext context)
        {
            BLegSession = context.Session;

....

With the UUID you can also call other API functions:

apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}",
Aleg.Uuid, BLegSession.GetUuid()));
apiResult = fsApi.Execute("uuid_media", "off " + call.Uuid);
apiResult = fsApi.ExecuteString(string.Format("uuid_kill {0}", call.Uuid));

Hope that helps


On Mon, Mar 8, 2010 at 9:30 PM, Chris <freeswitch at gilligan.id.au> wrote:

> That decently woks to originate the call but passing that UUID into a new
> ManagedSession does not seem to like to the session created using that UUID.
>  Once i create a session used the API you suggested how would i then access
> the session in the Managed Code?
>
> Chris
>
> On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> You could use the api:
>>
>>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
>> string apiResult = string.Empty;
>> string Uuid;
>> string NumberToDial = "3475558308";
>> string OutgoingCallerID = "2155556666";
>>
>> Uuid = fsApi.ExecuteString("create_uuid");
>>
>> apiResult = fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
>> Uuid, OutgoingCallerID, NumberToDial));
>>
>>
>>
>>
>>
>>
>> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>
>>> Hi,
>>> I am trying to create a mod_managed API application that takes 2 phone
>>> numbers as params.  These numbers should then be used to make 2 calls and to
>>> bridge them. The issue i am having is working out how to place the first
>>> call leg from in mod_managed.  All the methods i have found are for the
>>> second leg and require you to pass in the first call leg.
>>>
>>> I know there are easier way to do this out of the managed code but i want
>>> it in the managed code as this is just a proof of concept to prove we can do
>>> a callback like system via managed code to be hooked into other external
>>> systems.
>>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>>
>>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>>> while (s.ready()) {
>>> // The call has been answered
>>> }
>>>
>>> This seems to be exactly what i am looking for but seems to be missing
>>> from mod_managed. I am hoping someone can tell me how to do something
>>> similar in mod_managed since even if it is not part of the managed code
>>> mod_managed is meant to have the native access as well so i would assume it
>>> would be possible.
>>>
>>> Thanks in advance
>>>
>>> Chris
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wiltingtree at gmail.com  Tue Mar  9 18:26:06 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Tue, 9 Mar 2010 21:26:06 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>
Message-ID: <eb7797d71003091826j3fb9bb73k578aee8eed17a26e@mail.gmail.com>

So Anthony, don't take this the wrong way, and I'm sorry if I'm asking a
personal question, but if you've been writing this application for free for
the past 4 years, how do you support yourself?
Do you have a day job as well?


On Tue, Mar 9, 2010 at 3:43 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> This is hypochisy  in action.
>
> 99% of people asking questions in this ml are using FreeSWITCH for
> commercial use, I for one.
> So if the primary author of software that gets you paid asks for small
> bounty, I'd shut up or pay up.
>
>
> Ognjen
>
>
> On Tue, Mar 9, 2010 at 7:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I said I OFFER it in case of I finish my (no money no profit but big
>> potential) experimental project.
>> I understand perfectly your mind of how to manage an open source project,
>> but don't forget
>> that there are guys like you on this emailist who don't even make any
>> decent salary, that is why
>> I use and develop open source : share -> evovle your minds -> make all
>> people happy -> money comes
>> but not                                         share -> ask money -> make
>> people weird -> money won't come
>> it's my point of view.
>>
>> Regards
>>
>> ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 09, 2010 1:10 PM
>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> Aww, The les paul sounded nice I have one on my wishlist =D
>> http://bit.ly/bRHKlk
>>
>> *shrug* $250.00 maybe?
>>
>>
>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  OK How much ?
>>>
>>>  ----- Original Message -----
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> You would need to make a patch for it or pay someone to add it.
>>>
>>>
>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>> is not compatible)
>>>>
>>>> Thanks
>>>>
>>>> F
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Madovsky <infos at madovsky.org>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>> *Subject:* vars from sip_registrations table
>>>>
>>>> in sip_registration table there are
>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>
>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>> I need to replace the host of the callee in case of if he's registered
>>>> on another FS node.
>>>>
>>>> thanks
>>>>
>>>> Franck
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Mar  9 18:31:17 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Mar 2010 21:31:17 -0500
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
Message-ID: <367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>

One way might be to create an event loop and listen for DTMF

public bool Load()
        {
            ThreadPool.QueueUserWorkItem((o) =>
            {
                Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
                EventConsumer con = new EventConsumer("DTMF", "");
                while (true)
                {
                    Event ev = con.pop(1);
                    if (ev != null)
                    {
                                  ... do what ever needs to be done
                    }

                }
            });

See http://wiki.freeswitch.org/wiki/Event_List#DTMF

I am guessing the DTMF is "DTMF" so play around with that if it does not
work.

There are probably better ways.



On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:

> What i need to do is listen on a session before, during and after the
> session is bridged with another call. I know built into FS is the option to
> listen for *x where x is 0-9 but i really need more than one digit.
>
> I traced the method that listens for the single digit to the
> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
> switch_ivr_async.c
>
> mod_managed also has the following in the native class freeswitch.
> public static switch_status_t
> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
> recv_dtmf)
>
> The problem is i can't work out how to use it.  I am fine the the first
> param for the session but have no idea what i need for
> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
> recv_dtmf param.
>
> if anyone can point me in the right direction or knows a better way i would
> be grateful.
>
> Chris
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Mar  9 19:49:47 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 10 Mar 2010 11:49:47 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
Message-ID: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>

I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.

One running trunk. and back ported to another server running rev14696.
Sounds great.


2010/3/9 jay binks <jaybinks at gmail.com>:
> this is what I love about Freeswitch, and the FS development team.
> Well done Brian ( and the rest of the team ).
> Jay
>
>
> On Tue, Mar 9, 2010 at 11:07 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> We now have mod_silk in tree it has only been tested on Linux so far. ?The
>> silk library itself was released in the IETF draft only an hour and twenty
>> minutes ago. ?I wrote a perl script to extract the source from the draft.
>> ?Checked in the lib and the codec module which I had written to the binary
>> lib I had a few months ago.
>>
>> It has more work to do... but its there if anyone wants to chip in and
>> libtoolize the library, revamp the build system and assist in testing the
>> codec on multiple platforms.
>>
>> Thanks,
>> Brian
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Tue Mar  9 20:09:11 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 22:09:11 -0600
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
Message-ID: <5FA53946-544F-499B-9690-25E73FB5FA19@freeswitch.org>

Thanks.

/b

On Mar 9, 2010, at 9:49 PM, Seven Du wrote:

> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
> 
> One running trunk. and back ported to another server running rev14696.
> Sounds great.



From lloyd.aloysius at gmail.com  Tue Mar  9 20:59:14 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 9 Mar 2010 23:59:14 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
In-Reply-To: <20100310011738.GS18427@base.carmickle.com>
References: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
	<20100310011738.GS18427@base.carmickle.com>
Message-ID: <8a19bf2e1003092059o6895c8b3yceb8590d4f4be525@mail.gmail.com>

Frank ,

Thank you for your help.

Lloyd

On Tue, Mar 9, 2010 at 8:17 PM, Frank Carmickle <frank at carmickle.com> wrote:

> Hello
>
> On Tue, Mar 09, Aloysius Lloyd wrote:
> > Hi All,
> >
> > I am trying to setup a simple IVR with two options , timeout action and
> > Invalid digits pressed. here is the partial IVR Menu.
> >
> >
> > <entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
> > default"/>
> > <entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
> > default"/>
>
> Add something like.
>
>                timeout="3000"
>                inter-digit-timeout="2000"
>                max-failures="3"
>                max-timeouts="3"
>
> Then in the dialplan after you've called the ivr tranfer to another exten.
>  Be careful to make sure that hangup after bridge gets set so that you don't
> ring the exten when someone who has been called from the ivr hangs up.
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Tue Mar  9 21:10:23 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 10 Mar 2010 12:10:23 +0700
Subject: [Freeswitch-users] FS no frok mode
Message-ID: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>

Dear All,
               How about performance when start FS with -nf  (no
fork). I ask because i want to keep SQL connect (cash db) in  1
connection. but i want to know other effect like a performance ,
memory usage .. etc


Best Regrads.

Dome C.


From jason at jasonjgw.net  Tue Mar  9 21:13:44 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 10 Mar 2010 16:13:44 +1100
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
Message-ID: <20100310051344.GA19653@jdc.jasonjgw.net>

Seven Du <dujinfang at gmail.com> wrote:
> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
> 
> One running trunk. and back ported to another server running rev14696.
> Sounds great.

Do you have a comparison with my current favourite, CELT at 48000? CELT at 32000
also gives high quality, in my opinion.

I plan to test the SILK module at some point, possibly later this week.



From will.traenkle at yahoo.com  Tue Mar  9 21:23:07 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 21:23:07 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <20100308184025.GL18427@base.carmickle.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
	<20100308184025.GL18427@base.carmickle.com>
Message-ID: <51379.68073.qm@web57602.mail.re1.yahoo.com>

Frank,

Thanks for the tip.

I took your advice and added the following line to my huntgroup script:

session.execute("set","ignore_early_media=true");

and it works!

Thanks again,

-Will




________________________________
From: Frank Carmickle <frank at carmickle.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, March 8, 2010 10:40:25 AM
Subject: Re: [Freeswitch-users] hunt group - play music when trying external numbers

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC

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



      
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From dujinfang at gmail.com  Tue Mar  9 21:37:57 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 10 Mar 2010 13:37:57 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <20100310051344.GA19653@jdc.jasonjgw.net>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
Message-ID: <23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>

Unfortunately no.

I only tested 8000 because at least one leg is PSTN or SIP UA which
doesn't support SILK and CELT in our scenario. We are bridging UAs
between US and China so robustness is more concerned than HZ.

I just *feel* it's ok because I only tested with my ear without
considerate packet lost or jitter.

2010/3/10 Jason White <jason at jasonjgw.net>:
> Seven Du <dujinfang at gmail.com> wrote:
>> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
>>
>> One running trunk. and back ported to another server running rev14696.
>> Sounds great.
>
> Do you have a comparison with my current favourite, CELT at 48000? CELT at 32000
> also gives high quality, in my opinion.
>
> I plan to test the SILK module at some point, possibly later this week.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From will.traenkle at yahoo.com  Tue Mar  9 21:55:59 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 21:55:59 -0800 (PST)
Subject: [Freeswitch-users] IVR to hunt group to external destination fails
	// direct to hunt group to external desitnation succeeds
Message-ID: <153857.43476.qm@web57613.mail.re1.yahoo.com>

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.

However, if you access the hunt group directly without going through the main IVR, all works perfectly.  

Any ideas where I should look and what line of code I might be missing?

Thanks,

-Will



      
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From msc at freeswitch.org  Tue Mar  9 22:50:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 22:50:59 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003092250r6c32860ctde57e87563975f10@mail.gmail.com>

Hello all!

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

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

-Michael
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From msc at freeswitch.org  Tue Mar  9 23:00:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 23:00:53 -0800
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <153857.43476.qm@web57613.mail.re1.yahoo.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>

On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> When a hunt group is accessed from the main IVR, the hunt group dials an
> internal extension via transfer correctly but when trying to dial an
> external number such a cell phone via sip uri it is unable to connect or
> ring the phone but it waits the correct timeout period before it goes to
> voicemail.
>
> However, if you access the hunt group directly without going through the
> main IVR, all works perfectly.
>
> Any ideas where I should look and what line of code I might be missing?
>
>
Start by capturing the fs_cli debug output for the IVR call. Look through
the debug lines for clues as to what is happening with your call. The
symptom "unable to connect" has many possible causes, so try to locate the
debug lines leading up to the attempt to bridge the SIP URI. If you want
community help then pastebin the full debug log plus the relevant XML: the
dialplan extension and the IVR definition.
-MC
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From lloyd.aloysius at sunteltech.ca  Tue Mar  9 23:52:36 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 02:52:36 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly one device ring
Message-ID: <8a19bf2e1003092352r48acb809kb4bcd2c05d49eec1@mail.gmail.com>

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From lloyd.aloysius at gmail.com  Tue Mar  9 23:53:23 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 02:53:23 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
Message-ID: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From will.traenkle at yahoo.com  Wed Mar 10 00:39:41 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Wed, 10 Mar 2010 00:39:41 -0800 (PST)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
Message-ID: <474069.15451.qm@web57605.mail.re1.yahoo.com>

Michael,

Per your request:

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab 

If you require additional information or have any other questions, please let me know.

Thanks,

-Will



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:00:53 PM
Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds




On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>
>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>
>Any ideas where I should look and what line of code I might be missing?
>
>

Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
-MC


      
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From will.traenkle at yahoo.com  Wed Mar 10 00:45:40 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Wed, 10 Mar 2010 00:45:40 -0800 (PST)
Subject: [Freeswitch-users] auto attendant -> hunt group -> external
	number does not ring
In-Reply-To: <87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>
References: <302537.41702.qm@web57609.mail.re1.yahoo.com>
	<87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>
Message-ID: <533929.27849.qm@web57613.mail.re1.yahoo.com>

Michael,

There are two posts for this issue.  I just emailed the bottom links a few minutes ago, but reading your response prompted me to send one more link, which is the default.xml dialplan:

5) Dialplan default.xml: http://pastebin.com/iAbNSW9E 

Per
your request:

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab 
 
If you require additional information or have any other
questions, please let me know.

Thanks,

-Will



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:01:31 AM
Subject: Re: [Freeswitch-users] auto attendant -> hunt group -> external number does not ring




On Mon, Mar 8, 2010 at 2:23 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

>
>FS Community,
>
>
>I created an extension
>704
> 
>I created a
>hunt group 7004 that first rings 704 then rings my cell phone 16194548924.
> 
>I created an
>auto attendant 5002 with an option of 704 that goes to hunt group 7004.
> 
>1)    
>What does
>work: When I call the hunt group directly by dialing 7004 internally, the hunt
>group first rings 704 then rings my cell phone successfully.  This is perfect.
> 
>2)    
>What does not
>work: When I call the main auto attendant from either internally by dialing 5002
>or externally by dialing my main number, and dial 704, my extension rings and
>after not answering my cell phone does not ring.
> 
>What I am expecting
>in #2 is for my extension 704 to ring and if I do not pick up then my cell
>phone will ring just like it does in #1.
> 
>My knowledge
>is limited in this area and if you could point me in the right direction, that
>would be great.  I a have tried
>everything but with no luck.
>
>
Pastebin the necessary debug info. Check http://wiki.freeswitch.org/wiki/Reporting_Bugs for specifics on how to do collect and report all the data to pastebin. It would be good to see the dialplan and the debug log for each call. 

When you've collected all the data please reply here with the pastebin URL.
-MC


      
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From tculjaga at gmail.com  Wed Mar 10 01:42:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 10:42:49 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
Message-ID: <65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>

i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...

what is this related to ?

T.

On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:

> Hi Anthony,
>
> Many, many apologies to take so long to reply - I failed to notice your
> response.  Sorry !
>
> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
> then does not give a warning.
>
> Best wishes
> Bruce
>
>
> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> we are still wary about 5.3 due to bugs reported in libc.
>> What number does it say it detected for the gap?
>>
>>
>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>
>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>> ....
>>>
>>>
>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>
>>>> Hi,
>>>>
>>>> I wonder if anyone would be able to advise please:
>>>>
>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when
>>>> I start FreeSWITCH that
>>>>
>>>> "Abnormally large timer gap detected"
>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>> experience audio problems".
>>>>
>>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>>> to Centos5.4 was a yum update on the 5.3 build.
>>>>
>>>> I guess the warning comes from here:
>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>
>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>> optiplex p4's I use for testing.
>>>>
>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>> use such crummy hardware?!
>>>>
>>>> Many thanks in advance
>>>> Bruce
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Mar 10 01:51:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:51:18 -0500
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
Message-ID: <5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>

-nf has nothing to do with the performance, it has to do if the command blocks or not when you run it.  I am unsure what you mean about sql in this context, it is totally unrelated. What is a cash db? 

Mike


On Mar 10, 2010, at 12:10 AM, Dome Charoenyost wrote:

> Dear All,
>               How about performance when start FS with -nf  (no
> fork). I ask because i want to keep SQL connect (cash db) in  1
> connection. but i want to know other effect like a performance ,
> memory usage .. etc
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From mike at jerris.com  Wed Mar 10 01:54:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:54:53 -0500
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<4B8E6902.90106@fx-services.com>
	<d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>
Message-ID: <0E77A138-8D81-4A6D-BDB6-D94F25CD7B57@jerris.com>

I am not sure on this, if we are not using the same one here, I suspect that it is down in the sofia lib, and will require much more digging.  Try turing on ALL the sofia debug in the lib and see if any of the auth debug jumps out as to what might cause this.

Mike

On Mar 4, 2010, at 2:01 AM, Jonas Gauffin wrote:

> You are correct, I'm using DNSSRV. 
> 
> imho it's a bug when FS is not using the same nonce in the response as it received in the request. It might be a design flaw only that only appears when running against multiple servers through DNSSRV, but it's still a bug.
> 
> I'm sure it's easy to fix for Anthony with his m4d sk1llz ;)
> 



From mike at jerris.com  Wed Mar 10 01:56:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:56:33 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
In-Reply-To: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
	<282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
Message-ID: <82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>

curious what happens if you change the level or payload type int hat config below as well.

Mike

On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:

> Some more info . . .
>  
> The error only seems to occur when a "0" is entered.
>  
> Looking at the Phone config, the RFC2833 options are :-
>  
> In Band
> Out of Band (RFC2833) - currently selected
> SIP Info
>  
> RTP Payload Type (96-127) - 96 currently selected
> RTP DTMP Level (0-63) - 0 Currently selected
>  
> Dave
> ----- Original Message -----
> From: Dave Stevenson
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:23 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.
> 
> Hi Mathieu,
>  
> if you tell me how to generate one, I'll do it now ?
>  
> regards
> Dave
> ----- Original Message -----
> From: Mathieu Rene
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:09 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.
> 
> I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:
> 
>> Hi,
>>  
>> thanks Brian.
>>  
>> The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,
>>  
>> regards
>> Dave
>> ----- Original Message -----
>> From: Brian West
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Wednesday, March 03, 2010 7:50 PM
>> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.
>> 
>> It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.
>> 
>> /b
>> 
>> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>> 
>>>  
>>> It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?
>>>  
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Mar 10 02:00:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 05:00:20 -0500
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <43146.92621.qm@web44811.mail.sp1.yahoo.com>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
	<43146.92621.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <A13FDADE-EE1C-4B1E-A5E9-1EA547F2C201@jerris.com>

I notice that you properly indented this, so I guess this runs in python?


On Mar 4, 2010, at 4:10 PM, Jim Thomas wrote:

> > PYTHON SUCKS
>  
> That's like saying:
>  
>   Asterisk sucks.
>   Asterisk is written in C language.
>   Therefore, C language sucks.
>  
> :-)
> 
>  
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, March 04, 2010 9:52 AM
> Subject: [Freeswitch-users] SUCCESS!!!
> 
> 
> >I have to say PYTHON SUCKS... ie Mailman is evil.
> > 

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From mike at jerris.com  Wed Mar 10 02:01:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 05:01:46 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
	<7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
Message-ID: <F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>

What pri dialect is this?

On Mar 4, 2010, at 11:11 PM, lakshmanan ganapathy wrote:

> Ok. This is to say how the problem got solved.
> Need openzap 1047 or above version.
> Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz 
> My telco is not accepting the display IE. 
> Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.
> 
> Thanks for all.
> 
> On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn <tomek.augustyn at gmail.com> wrote:
> Hello,
> 
> I had similar problem and I think it is more a problem between Sangoma card and your E1 provider than with freeswitch.
> 
> In my case it was necessary to set "origination_caller_id_number" to one of the telephone numbers linked to my E1 line. In other case the calls were rejected with "invalid information element" error.
> 
> You can try Sangoma's support they are very helpful.
> 
> Tomasz Augustyn

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From dome at tel.co.th  Wed Mar 10 02:12:58 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 10 Mar 2010 17:12:58 +0700
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
	<5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
Message-ID: <8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>

2010/3/10 Michael Jerris <mike at jerris.com>:
> -nf has nothing to do with the performance, it has to do if the command
> blocks or not when you run it. ?I am unsure what you mean about sql in this
> context, it is totally unrelated. What is a cash db?
sorry i mean switch_cache_db_connection :)
i have problem about db connection. now switch_cache_db_connection use
1 connection for 1 thread
if i use mod_limit with odbc my server can't handle more than 100 connection.
So i'll try FS in no fork. but i want to know other effect.

BG

Dome C.
> Mike
>
> On Mar 10, 2010, at 12:10 AM, Dome Charoenyost wrote:
>
> Dear All,
> ??????????????How about performance when start FS with -nf ?(no
> fork). I ask because i want to keep SQL connect (cash db) in ?1
> connection. but i want to know other effect like a performance ,
> memory usage .. etc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jbrucehopkins at gmail.com  Wed Mar 10 04:00:31 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 12:00:31 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <000001cabfbb$f2114c90$d633e5b0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
Message-ID: <d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>

Hi Roly,

I hope to be able to have a go at this later today.  I'll let you know what
happens.

I have my FS box behind a hardware router/firewall, so I'm assuming I just
forward SIP port 5060 from the WAN to the FS box and use internal profile,
instead of using external profile with port 5080 forwarded as I have at the
moment.  We'll see what happens.

cheers
Bruce

On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Wed Mar 10 04:02:39 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Wed, 10 Mar 2010 23:02:39 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com> 
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
Message-ID: <9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>

Thanks for the idea i will try that next. I thought i would try to capture
the digits like normal first to make sure everything is working right and to
test a few event capture methods.  What i found was that mod_managed never
seemed to capture or read any digits for an out going call.  If i pass the
call to a different application/dial plan it captures the digits perfectly.
 I assume audio was working as it played audio to the user to enter digits
so i figure i have not configured something correctly but i have not idea
what it is.  Until i work it out i don't think any event code will work.

I have no idea what is wrong in this case or even how to debug it.

Chris

On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> One way might be to create an event loop and listen for DTMF
>
> public bool Load()
>         {
>             ThreadPool.QueueUserWorkItem((o) =>
>             {
>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>                 EventConsumer con = new EventConsumer("DTMF", "");
>                 while (true)
>                 {
>                     Event ev = con.pop(1);
>                     if (ev != null)
>                     {
>                                   ... do what ever needs to be done
>                     }
>
>                 }
>             });
>
> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>
> I am guessing the DTMF is "DTMF" so play around with that if it does not
> work.
>
> There are probably better ways.
>
>
>
> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> What i need to do is listen on a session before, during and after the
>> session is bridged with another call. I know built into FS is the option to
>> listen for *x where x is 0-9 but i really need more than one digit.
>>
>> I traced the method that listens for the single digit to the
>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>> switch_ivr_async.c
>>
>> mod_managed also has the following in the native class freeswitch.
>> public static switch_status_t
>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>> recv_dtmf)
>>
>> The problem is i can't work out how to use it.  I am fine the the first
>> param for the session but have no idea what i need for
>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>> recv_dtmf param.
>>
>> if anyone can point me in the right direction or knows a better way i
>> would be grateful.
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Wed Mar 10 04:07:32 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 13:07:32 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <65d96fc81003100407o4c55b3f3y1ba35c6fcfe5dc66@mail.gmail.com>

On Mon, Mar 8, 2010 at 11:02 PM, Brian West <brian at freeswitch.org> wrote:

> Thats unacceptable!!!  Wait its faxing... can't we just let it DIE
> already??  They have this thing called email... you should check it out...
> works great... even lets you attach video, audio and color images.... oh and
> text... dreary old text.... but hey you can mark it all up in HTML and fancy
> fonts now and make it anoying^H^H^H^H^H^H^Hpretty.
>
> /b
>
>
well Brian, thats a wrong assumption.... fax (T.30) is, and it will be,
present all over the world... you cannot do anything against (not even thing
about eradicate) it at least for the next 5 years :)

so, you will ave to comply ... we will have to comply if our customers keep
sending faxes :)


T.
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From tculjaga at gmail.com  Wed Mar 10 04:10:57 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 13:10:57 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
Message-ID: <65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>

they just want to believe it is :) .. thats it.


T.

On Mon, Mar 8, 2010 at 11:46 PM, Brian West <brian at freeswitch.org> wrote:

> There is no such thing as security when humans are involved in the relay of
> information.  Thats fact.
>
> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
> involved its not secure.
>
> /b
>
> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>
> > While I'd love to see FAXing just die, especially with the amount of
> headache associated with it, the reality is that specific organizations
> still rely on it heavily on a daily basis (ie Health Care and
> Banking/Finance Industry). Plus they can legally get away with FAXing
> personal data directly between machines, since the FAX transmission doesn't
> seem to be directly covered by all their respective privacy regultions (ie
> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
> of these organizations just don't have the internal expertise to setup
> end-to-end encrypted email nor the impetus to deal with it, since FAXing
> works fine in their eyes. I think it's going to be around quite a while
> longer unless they actually stop making the FAX machines and force people to
> use something else.
> >
> > Cheers,
> > Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Wed Mar 10 05:42:34 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 10 Mar 2010 08:42:34 -0500
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
	<9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>
Message-ID: <367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>

There should be no problem on digit capture on an outgoing call. For example
- when I originate a call, I ask the called party, for example, to press 1
to accept the call. This works correctly:

BLegSession.DtmfReceivedFunction = (d, t) =>
                {
                    Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d,
t);
                    if (d == confirmationDigit)
                    {
                        confirmed = true;
                        return "break";
                    }

                    return "";
                };

                while (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady() && loop <= loops)
                {
                    loop++;

                    if (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady())

BLegSession.StreamFile("prompts/press-1-to-accept-call-from.wav", 0);

                    if (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady())
                        BLegSession.CollectDigits(5000);

                }

If you are not receiving the digits - I would try and bridge the oubound
call to an existing call and check that you are getting two way audio - and
if you are using inband DTMF -  that you can actually hear the digits.

HTH


On Wed, Mar 10, 2010 at 7:02 AM, Chris <freeswitch at gilligan.id.au> wrote:

> Thanks for the idea i will try that next. I thought i would try to capture
> the digits like normal first to make sure everything is working right and to
> test a few event capture methods.  What i found was that mod_managed never
> seemed to capture or read any digits for an out going call.  If i pass the
> call to a different application/dial plan it captures the digits perfectly.
>  I assume audio was working as it played audio to the user to enter digits
> so i figure i have not configured something correctly but i have not idea
> what it is.  Until i work it out i don't think any event code will work.
>
> I have no idea what is wrong in this case or even how to debug it.
>
> Chris
>
>
> On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> One way might be to create an event loop and listen for DTMF
>>
>> public bool Load()
>>         {
>>             ThreadPool.QueueUserWorkItem((o) =>
>>             {
>>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>>                 EventConsumer con = new EventConsumer("DTMF", "");
>>                 while (true)
>>                 {
>>                     Event ev = con.pop(1);
>>                     if (ev != null)
>>                     {
>>                                   ... do what ever needs to be done
>>                     }
>>
>>                 }
>>             });
>>
>> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>>
>> I am guessing the DTMF is "DTMF" so play around with that if it does not
>> work.
>>
>> There are probably better ways.
>>
>>
>>
>> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>
>>> What i need to do is listen on a session before, during and after the
>>> session is bridged with another call. I know built into FS is the option to
>>> listen for *x where x is 0-9 but i really need more than one digit.
>>>
>>> I traced the method that listens for the single digit to the
>>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>>> switch_ivr_async.c
>>>
>>> mod_managed also has the following in the native class freeswitch.
>>> public static switch_status_t
>>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>> recv_dtmf)
>>>
>>> The problem is i can't work out how to use it.  I am fine the the first
>>> param for the session but have no idea what i need for
>>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>> recv_dtmf param.
>>>
>>> if anyone can point me in the right direction or knows a better way i
>>> would be grateful.
>>>
>>> Chris
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ahmed.ajmal at breezecom.ae  Wed Mar 10 06:01:38 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Wed, 10 Mar 2010 19:01:38 +0500
Subject: [Freeswitch-users] Set/Export channel variable
Message-ID: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>

Hi 

 

I am trying to set a channel variable on the bridge application using this:

 

<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>

 

The application runs fine, what I now need to do is set/export the 'gw'
channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.

 

Thanks

Ahmed

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From brian at freeswitch.org  Wed Mar 10 06:08:25 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Mar 2010 08:08:25 -0600
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
References: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
Message-ID: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>

We already set you a variable called sip_gateway_name when a call is placed out a gateway.  You're also doing this wrong you should NEVER EVER do @$ipgwX as that data is in the gateway.  You should simply do sofia/gateway/gwname/number

/b

On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:

> Hi
>  
> I am trying to set a channel variable on the bridge application using this:
>  
> <action application="bridge" data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 | [gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>
>  
> The application runs fine, what I now need to do is set/export the ?gw? channel variable for CDR so that I know which gateway was dialed. I have tried using set and export application but that doesn?t work. Please help.
>  
> Thanks
> Ahmed

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From srinivas.ksvreddy at gmail.com  Wed Mar 10 06:39:45 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 10 Mar 2010 09:39:45 -0500
Subject: [Freeswitch-users] forwarding incoming call from gateway to local
	externsion
Message-ID: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>

Hi Goodmorning.

i have connected my freeswtich to cisco gateway, i have registerd a
user(1099) in my freeswitch, and i have configured freeswitch any call from
1099 routing to cisco gateway, i need same way when call coming from gateway
with 101010 i want to route or farward to 1099, what files need to configure
any idea

Regard
srinvias
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From anthony.minessale at gmail.com  Wed Mar 10 08:25:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 10:25:46 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
Message-ID: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>

FS now tries to determine how accurate the timing is by doing some tests
with 1ms sleeps.
Most 64 bit CentOS box I have seen can easily calibrate.
If you are having that error alot but the calls still sound ok, it's safe to
ignore it.
you can also start FS with -nocal to skip this test.

Also make sure you do not have any other cpu-intensive application on your
box possibly competing for resources.


On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>
> what is this related to ?
>
> T.
>
>
> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>
>> Hi Anthony,
>>
>> Many, many apologies to take so long to reply - I failed to notice your
>> response.  Sorry !
>>
>> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
>> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
>> then does not give a warning.
>>
>> Best wishes
>> Bruce
>>
>>
>> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>>
>>> we are still wary about 5.3 due to bugs reported in libc.
>>> What number does it say it detected for the gap?
>>>
>>>
>>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com
>>> > wrote:
>>>
>>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>>> ....
>>>>
>>>>
>>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I wonder if anyone would be able to advise please:
>>>>>
>>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when
>>>>> I start FreeSWITCH that
>>>>>
>>>>> "Abnormally large timer gap detected"
>>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>>> experience audio problems".
>>>>>
>>>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>>>> to Centos5.4 was a yum update on the 5.3 build.
>>>>>
>>>>> I guess the warning comes from here:
>>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>>
>>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>>> optiplex p4's I use for testing.
>>>>>
>>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>>> use such crummy hardware?!
>>>>>
>>>>> Many thanks in advance
>>>>> Bruce
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 10 09:09:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 11:09:46 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
	<alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
Message-ID: <191c3a031003100909t37e97e92r2729dc6470d7e6d6@mail.gmail.com>

you can change the delimiter

Like the example:

"show calls as csv :::"

The data will be sep by ::: instead of comma

if you would like to make a patch to add "" it would be applied.



On Mon, Mar 1, 2010 at 9:29 PM, Wellie Chao <wchao at yahoo.com> wrote:

> Terrific. "show calls as xml" works great and just what I needed.
>
> P.S. "show calls as csv" still has the comma bug. Not a big deal since I
> can get it as XML. CSV needs quotes around field values that contain a
> comma. Pretty easy to fix. Few lines of C (strchr for "," and if found,
> put quotes around field value).
>
>
> Date: Mon, 1 Mar 2010 17:24:16 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Bug in mod_commands.c with show calls
> command
>
> Perhaps you may be reading up on something quickly and not worrying about
> the absolute correctness of something that is not critical to our time to
> explain.
>
> The proper syntax for the command is "show calls as xml" and you are not
> really using XMLRPC you are hitting it via direct url.
> so you will then need.
>
> http://192.168.1.1:8080/txtapi/show?calls%20as%20xml
>
>
> anything in the show command can be replied to with "as xml"
>
> also, for fun, try
>
> show calls as csv :::
>
>
>
>
> On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
>      I am using the xml_rpc interface like so:
>
>      http://192.168.1.1:8080/webapi/show?calls
>
>      I get a response in a table, which is a little bit undesirable because
> I
>      want to parse the result in a program and don't want to parse a bunch
> of
>      HTML (just seems inelegant and wasteful), so I instead tried the
> following
>      request:
>
>      http://192.168.1.1:8080/txtapi/show?calls
>
>      That works better, but then another wrinkle appears. Caller ID (and
> Callee
>      ID) names sometimes have commas, which messes up the CSV.
>
>      This appears to be a bug in mod_commands.c (or I suppose you could
> call it
>      an artifact of somebody coding up something quickly and not worrying
> about
>      the absolute correctness of something that is not critical to the
> core).
>      There is also another bug with show calls via webapi in that the
> generated
>      HTML has spurious <td> tags. I think the spurious <td> tag bug arises
> due
>      to lines 3145-6 in mod/applications/mod_commands/mod_commands.c.
> Finally,
>      another bug (or maybe just unimplemented feature) is that xmlapl isn't
>      really different from webapi.
>
>      These problems occur in the latest source tree (20100301).
>
>      I am wondering if (a) one of the developers can fix these or (b) if I
> fix
>      the bugs, how can I submit patches [and if I submit patches, will they
> be
>      accepted into the main tree]?
>
>      The fixes are pretty trivial and I'd be happy to code them up if
> somebody
>      will tell me how I can submit the patches (haven't done it before).
>
>      _______________________________________________
>      FreeSWITCH-users mailing list
>      FreeSWITCH-users at lists.freeswitch.org
>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>      http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 10 09:11:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 11:11:41 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
Message-ID: <191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>

We have a mod_fax for t30 and work on t38 is underway.
So we already acknowledge that I guess.




On Wed, Mar 10, 2010 at 6:10 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> they just want to believe it is :) .. thats it.
>
>
> T.
>
>
> On Mon, Mar 8, 2010 at 11:46 PM, Brian West <brian at freeswitch.org> wrote:
>
>> There is no such thing as security when humans are involved in the relay
>> of information.  Thats fact.
>>
>> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
>> involved its not secure.
>>
>> /b
>>
>> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>>
>> > While I'd love to see FAXing just die, especially with the amount of
>> headache associated with it, the reality is that specific organizations
>> still rely on it heavily on a daily basis (ie Health Care and
>> Banking/Finance Industry). Plus they can legally get away with FAXing
>> personal data directly between machines, since the FAX transmission doesn't
>> seem to be directly covered by all their respective privacy regultions (ie
>> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
>> of these organizations just don't have the internal expertise to setup
>> end-to-end encrypted email nor the impetus to deal with it, since FAXing
>> works fine in their eyes. I think it's going to be around quite a while
>> longer unless they actually stop making the FAX machines and force people to
>> use something else.
>> >
>> > Cheers,
>> > Gabe
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From chris at fowler.cc  Wed Mar 10 09:16:10 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 10 Mar 2010 12:16:10 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only
	one	device ring
In-Reply-To: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>

Hi Lloyd,

You need to use sofia_contact in your bridge command in the dialplan:

e.g.
<anti-action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

This will cause FS to try all registrations simultaneously.

Cheers, Chris.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius Lloyd
Sent: Tuesday, March 09, 2010 11:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From stevendt at primrosebank.net  Wed Mar 10 09:44:03 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 10 Mar 2010 17:44:03 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>
Message-ID: <C1A1AC0759924ED19A5FD2ABF82A9D2E@bp1.ad.bp.com>


  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 10, 2010 9:56 AM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  curious what happens if you change the level or payload type int hat config below as well.


  Mike


  On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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



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From stevendt at primrosebank.net  Wed Mar 10 09:50:24 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 10 Mar 2010 17:50:24 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>
Message-ID: <B8B7700F9FBA4CE3B1B6F059307C88CE@bp1.ad.bp.com>

Mike,

thanks for following this up . . . . .

Not having the faintest idea what those settings might do, I never thought of changing them and see what difference it made, but you prompted me to try that....


I've still no idea what "RTP Payload Type" is, but changing it makes no difference to my problem.

However, DTMF level (the "DTMP" was a typo in my earlier message), which I'm guessing is a kind of volume setting (?) DOES make a difference.

Setting it to different values makes the error go away, it's currently set at "1" (from "0") which seems to have been enough to get rid of the problem. Strange that the problem only affected one digit ("0"), but maybe something was just on a limit somewhere - maybe the phone has an issue with certain frequencies.

Anyway, a simple work around, if only I'd had the sense to try that earlier - thanks a lot for the prompt !

regards
Dave


  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 10, 2010 9:56 AM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  curious what happens if you change the level or payload type int hat config below as well.


  Mike


  On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From peder at networkoblivion.com  Wed Mar 10 11:27:03 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 10 Mar 2010 13:27:03 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <019801cac087$a9da3b90$fd8eb2b0$@com>

Is there a way to tell how accurate the timing is on a box before running
FreeSWITCH on it?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, March 10, 2010 10:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4

 

FS now tries to determine how accurate the timing is by doing some tests
with 1ms sleeps.
Most 64 bit CentOS box I have seen can easily calibrate.
If you are having that error alot but the calls still sound ok, it's safe to
ignore it.
you can also start FS with -nocal to skip this test.

Also make sure you do not have any other cpu-intensive application on your
box possibly competing for resources.



On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...

what is this related to ?

T.

 

On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
wrote:

Hi Anthony,

Many, many apologies to take so long to reply - I failed to notice your
response.  Sorry !

It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
then does not give a warning.

Best wishes
Bruce

 

On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

we are still wary about 5.3 due to bugs reported in libc.
What number does it say it detected for the gap?



On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
wrote:

OK - I've realised I do get the same warning with CentOS 5.3, it just goes
past more quickly so I didn't see it.  Maybe it is just the hardware ....

 

On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

Hi,

I wonder if anyone would be able to advise please:

When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
start FreeSWITCH that 

"Abnormally large timer gap detected"
"Do you have your kernel timer set to greater than 1kHz?  You may experience
audio problems".

I get no such warning if I build on CentOS 5.3, and the test timings it
measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
to Centos5.4 was a yum update on the 5.3 build.

I guess the warning comes from here:
http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time
.c
<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_t
ime.c> 

This is all on pretty low spec hardware - a couple of different Dell
optiplex p4's I use for testing.

Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
and not worry about the warnings, or if there is something I can do to fix
the problem it is warning about.  Perhaps it is just that I shouldn't use
such crummy hardware?!

Many thanks in advance
Bruce

 

 

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






-- 
Anthony Minessale II

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

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

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


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

 


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

 


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




-- 
Anthony Minessale II

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

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

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

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From jbrucehopkins at gmail.com  Wed Mar 10 11:38:21 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 19:38:21 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <d03a10e01003101138n65918b8bx6798d0860a982d12@mail.gmail.com>

Many thanks for the info.  I was using a fairly crummy 32 bit box with no
PSTN card installed.  There was no degredation of the audio immediately
apparent.

Bests
Bruce

On 10 March 2010 16:25, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>
>
> On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>>
>> what is this related to ?
>>
>> T.
>>
>>
>> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>
>>> Hi Anthony,
>>>
>>> Many, many apologies to take so long to reply - I failed to notice your
>>> response.  Sorry !
>>>
>>> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
>>> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
>>> then does not give a warning.
>>>
>>> Best wishes
>>> Bruce
>>>
>>>
>>> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>>>
>>>> we are still wary about 5.3 due to bugs reported in libc.
>>>> What number does it say it detected for the gap?
>>>>
>>>>
>>>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <
>>>> jbrucehopkins at gmail.com> wrote:
>>>>
>>>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>>>> ....
>>>>>
>>>>>
>>>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I wonder if anyone would be able to advise please:
>>>>>>
>>>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning
>>>>>> when I start FreeSWITCH that
>>>>>>
>>>>>> "Abnormally large timer gap detected"
>>>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>>>> experience audio problems".
>>>>>>
>>>>>> I get no such warning if I build on CentOS 5.3, and the test timings
>>>>>> it measures on starting FreeSWITCH do look lower.  All I was doing to
>>>>>> upgrade to Centos5.4 was a yum update on the 5.3 build.
>>>>>>
>>>>>> I guess the warning comes from here:
>>>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>>>
>>>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>>>> optiplex p4's I use for testing.
>>>>>>
>>>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or
>>>>>> use 5.4 and not worry about the warnings, or if there is something I can do
>>>>>> to fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>>>> use such crummy hardware?!
>>>>>>
>>>>>> Many thanks in advance
>>>>>> Bruce
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 10 11:44:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Mar 2010 11:44:43 -0800
Subject: [Freeswitch-users] forwarding incoming call from gateway to
	local externsion
In-Reply-To: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>
References: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>
Message-ID: <87f2f3b91003101144w31dc79b3x27c1d5e5d6f8b449@mail.gmail.com>

On Wed, Mar 10, 2010 at 6:39 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi Goodmorning.
>
> i have connected my freeswtich to cisco gateway, i have registerd a
> user(1099) in my freeswitch, and i have configured freeswitch any call from
> 1099 routing to cisco gateway, i need same way when call coming from gateway
> with 101010 i want to route or farward to 1099, what files need to configure
> any idea
>

Look in conf/dialplan/public.xml and see how the default extension numbers
(1000-1019) are handled in this scenario.
-MC
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From anthony.minessale at gmail.com  Wed Mar 10 12:29:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 14:29:02 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <019801cac087$a9da3b90$fd8eb2b0$@com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
	<019801cac087$a9da3b90$fd8eb2b0$@com>
Message-ID: <191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>

The best tool we have is actually running freeswitch on it to see.


On Wed, Mar 10, 2010 at 1:27 PM, Peder <peder at networkoblivion.com> wrote:

>  Is there a way to tell how accurate the timing is on a box before running
> FreeSWITCH on it?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, March 10, 2010 10:26 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4
>
>
>
> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>  On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>
> what is this related to ?
>
> T.
>
>
>
> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
> wrote:
>
> Hi Anthony,
>
> Many, many apologies to take so long to reply - I failed to notice your
> response.  Sorry !
>
> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
> then does not give a warning.
>
> Best wishes
> Bruce
>
>
>
> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> we are still wary about 5.3 due to bugs reported in libc.
> What number does it say it detected for the gap?
>
>   On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
> wrote:
>
>  OK - I've realised I do get the same warning with CentOS 5.3, it just
> goes past more quickly so I didn't see it.  Maybe it is just the hardware
> ....
>
>
>
> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
> Hi,
>
> I wonder if anyone would be able to advise please:
>
> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
> start FreeSWITCH that
>
> "Abnormally large timer gap detected"
> "Do you have your kernel timer set to greater than 1kHz?  You may
> experience audio problems".
>
> I get no such warning if I build on CentOS 5.3, and the test timings it
> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
> to Centos5.4 was a yum update on the 5.3 build.
>
> I guess the warning comes from here:
> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>
> This is all on pretty low spec hardware - a couple of different Dell
> optiplex p4's I use for testing.
>
> Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
> and not worry about the warnings, or if there is something I can do to fix
> the problem it is warning about.  Perhaps it is just that I shouldn't use
> such crummy hardware?!
>
> Many thanks in advance
> Bruce
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From siniypin at gmail.com  Wed Mar 10 12:52:00 2010
From: siniypin at gmail.com (RobertT)
Date: Wed, 10 Mar 2010 23:52:00 +0300
Subject: [Freeswitch-users] sofia_contact strange behavior
Message-ID: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>

Hello everybody!

I am experiencing problems with sofia_contact command.
When I dial sofia_contact external/1000@<domain> in fs cli I see correct
registration data, provided the user 1000 is registered.
But in the same time when I call
${sofia_conatct(external/${dialed_ext}@$${domain})}
from dialplan as an argument to deflect application I recieve
error\user_not_registered.
What the hell?

Best regards, RobertT.
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From peder at networkoblivion.com  Wed Mar 10 12:53:47 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 10 Mar 2010 14:53:47 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
Message-ID: <021b01cac093$c76d3930$5647ab90$@com>

OK, can you decode what this means exactly?  I'd be happy to write something
up to reference users to, but I don't really know what it means.  It is a a
quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
150, but is that good, bad, indifferent?  I've tried to google it, but
without really knowing what it is I am looking for, I am not getting good
results.

 

 

2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating timer,
please wait...

2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000 Average:
2001 Step: 50

2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
1998 Step: 50

2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
1998 Step: 50

2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
1998 Step: 50

2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
1998 Step: 50

2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
1998 Step: 50

2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
1998 Step: 50

2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
1997 Step: 50

2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
1998 Step: 50

2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
1998 Step: 50

2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
1998 Step: 50

2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
1998 Step: 50

2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
1998 Step: 50

2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
1018 Step: 50

2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
1018 Step: 50

2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
1018 Step: 50

2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
1038 Step: 50

2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
1018 Step: 50

2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
1018 Step: 50

2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
1018 Step: 50

2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average: 1018
Step: 50

2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000 Average:
2018 Step: 50

2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
1998 Step: 50

2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
1578 Step: 50

2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of 150
calculated

2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [CORE_SOFTTIMER_MODULE]

2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
Timer 'soft'

2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [CORE_PCM_MODULE]

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From ben at langfeld.co.uk  Wed Mar 10 13:05:32 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 10 Mar 2010 21:05:32 +0000
Subject: [Freeswitch-users] Tone Detection During Bridge
Message-ID: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>

Dear List,

I have a requirement (due to poor quality Sipura products) to have
freeswitch listen for a tone during a bridge (standard local extension
dialplan segment). I have the required line:

<action appication="tone_detect" data="hangup 400 r 0 hangup
'normal_clearing' 1"/>

But I need to have this run _during_ the bridge (to listen for the tone
coming from the broken SPA3k and to break the bridge). Does anyone have any
idea how this may be achieved? There is no such documentation on the wiki.
This works perfectly if the bridge was not successful, and so will detect
tones during voicemail messaging.

Regards,
Ben Langfeld

Wave > Email
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From ledoktre at meanie.us  Wed Mar 10 13:26:15 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 10 Mar 2010 15:26:15 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
	<4B8569CB.6070804@meanie.us>
	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
Message-ID: <4B980E77.8080303@meanie.us>

Giovanni,

Any progress updates on the skypiax updates for sound quality and also 
DMTF detection?

With this update coming, I am pretty excited to be trying it out.  I 
have been trying very, very hard to be patient... :-D

Thanks,

Giovanni Maruzzelli wrote:
> On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre at meanie.us> wrote:
>   
>> Hello,
>>
>> I was writing to inquire how this skypiax update was coming along.  I
>> didn't see it in the mailing list, but also since it had been nearly 1
>> week, I wanted to be sure if I didn't miss the announcement.  I am very
>> anxious to try this new update of the module.. It should be awesome!
>>     
>
> Hehehe, no, you've not missed the announcement.
>
> Is taking me some more time than I was expecting.
> But's arriving...
> I'll post here the announcement ;)
>
> -giovanni
>
>
>
>   
>> Thanks,
>>
>> Tim
>>
>> Giovanni Maruzzelli wrote:
>>     
>>> before to delve in the troubleshooting, I have to say that I'm
>>> modifying the audio skypiax code in svn, so maybe it's just my fault
>>> ;).
>>>
>>> please be patient for a little while, I hope to have done with it in a
>>> couple days.
>>>
>>> I'll announce to the mailing list when done.
>>>
>>> In the mean time, at least one good news for you user of SkypeIn
>>> service: a new feature of mod_skypiax is intended to recognize the
>>> DTMFs coming from SkypeIn, so the incoming calls will be able to use
>>> ivr, voicemail, etc.
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   


From jbrucehopkins at gmail.com  Wed Mar 10 13:28:30 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 21:28:30 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
Message-ID: <d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>

Hey Roly,

Are you getting anywhere?  I'm still struggling, I'm afraid.

I'm able to call from WAN --> LAN using the internal profile on port 5060,
but I still can't call out from LAN --> WAN so there is obviously something
really basic I'm still not getting.

In sip_profiles/internal.xml I have:

 <param name="aggressive-nat-detection" value="true"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>

<!-- the external_rtp_ip etc being defined in vars.xml -->


In this example I have left <param name="force-register-domain"
value="$${domain}"/> etc uncommented to force the registration to go to
default.xml in the dialplan.

I thought maybe this was my problem and tried putting a separate file in the
directory to enable the WAN phone to register using the external DNS name
for the server as its domain, and uncommenting force-register-domain, etc.
But then I got an error saying that the local (to Freeswitch) phone in the
LAN was not registered when I tried to make the call from the WAN.

Any thoughts?

cheers
Bruce






On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi Roly,
>
> I hope to be able to have a go at this later today.  I'll let you know what
> happens.
>
> I have my FS box behind a hardware router/firewall, so I'm assuming I just
> forward SIP port 5060 from the WAN to the FS box and use internal profile,
> instead of using external profile with port 5080 forwarded as I have at the
> moment.  We'll see what happens.
>
> cheers
> Bruce
>
> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>
>> Ah I see... what? Please share and lead this blind man out the FS
>> wilderness!
>>
>> I don't understand...what happens to the external profile? Do you delete
>> it?
>> And how do you forward port 5060?
>>
>> ...and you thought you were a newbie!
>>
>> Any insight would be much appreciated...loving the journey.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
>> Hopkins
>> Sent: 09 March 2010 08:23 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>
>> Ah I see.  I will try again using the internal profile and forwarding
>> port 5060.  Presumably still creating a directory entry to enable the
>> outside-facing domain to be used.
>>
>> Many thanks for your patient help of a newbie Brian.
>>
>> Cheers
>> Bruce
>>
>> Please excuse the brevity - sent from my mobile.
>>
>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>
>> > ok you aren't catching one key thing here.. you no longer need two
>> > profiles.
>> >
>> > /b
>> >
>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>> >
>> >> Hi again,
>> >>
>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>> >> something wrong here as I am still having difficulty, and still
>> >> have the same problem.
>> >>
>> >> I updated to build 16938 by means of "make current"
>> >>
>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>> >> to use 5080 instead of 5090.
>> >>
>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>> >> the server as a domain for the remote softphone to register on.  I
>> >> left the group name in this entry the same as inthe default entry,
>> >> so that the remote phone could register on the same extension
>> >> numbers (100, etc) as in the default build.
>> >>
>> >> I still find that, if I initiate a call from the local (on same LAN
>> >> as freeswitch) phone to the remote phone, I get the message on the
>> >> CLI:
>> >>
>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>> >> type [user] cause: [USER_NOT_REGISTERED]
>> >>
>> >> One possibly unrelated aside, I also found I needed to uncomment
>> >> <param name="aggressive-nat-detection" value="true"/> in
>> >> external.xml, otherwise in the case of a call initiated by the
>> >> remote phone being hung up by the local phone, freeswitch sent the
>> >> BYE to the private IP of the remote phone, rather than its public
>> >> ip - meaning that the remote phone didn't receive the BYE.
>> >>
>> >> Any further ideas where I am going wrong here please?
>> >>
>> >> thanks again in advance
>> >> Bruce
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From freeswitch at gilligan.id.au  Wed Mar 10 13:42:40 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Thu, 11 Mar 2010 08:42:40 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com> 
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com> 
	<9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com> 
	<367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>
Message-ID: <9394d42f1003101342p55c26227xdeb298dec3f68d11@mail.gmail.com>

Thanks Phillip! That worked perfectly for some reason.  I mean it was hardly
different from mine and i still can't spot the fault in my own code.  very
strange.


On Thu, Mar 11, 2010 at 12:42 AM, Phillip Jones <pjintheusa at gmail.com>wrote:

> There should be no problem on digit capture on an outgoing call. For
> example - when I originate a call, I ask the called party, for example, to
> press 1 to accept the call. This works correctly:
>
> BLegSession.DtmfReceivedFunction = (d, t) =>
>                 {
>                     Log.WriteLine(LogLevel.Info, "Received {0} for {1}.",
> d, t);
>                     if (d == confirmationDigit)
>                     {
>                         confirmed = true;
>                         return "break";
>                     }
>
>                     return "";
>                 };
>
>                 while (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady() && loop <= loops)
>                 {
>                     loop++;
>
>                     if (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady())
>
> BLegSession.StreamFile("prompts/press-1-to-accept-call-from.wav", 0);
>
>                     if (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady())
>                         BLegSession.CollectDigits(5000);
>
>                 }
>
> If you are not receiving the digits - I would try and bridge the oubound
> call to an existing call and check that you are getting two way audio - and
> if you are using inband DTMF -  that you can actually hear the digits.
>
> HTH
>
>
>
> On Wed, Mar 10, 2010 at 7:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Thanks for the idea i will try that next. I thought i would try to capture
>> the digits like normal first to make sure everything is working right and to
>> test a few event capture methods.  What i found was that mod_managed never
>> seemed to capture or read any digits for an out going call.  If i pass the
>> call to a different application/dial plan it captures the digits perfectly.
>>  I assume audio was working as it played audio to the user to enter digits
>> so i figure i have not configured something correctly but i have not idea
>> what it is.  Until i work it out i don't think any event code will work.
>>
>> I have no idea what is wrong in this case or even how to debug it.
>>
>> Chris
>>
>>
>> On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>> One way might be to create an event loop and listen for DTMF
>>>
>>> public bool Load()
>>>         {
>>>             ThreadPool.QueueUserWorkItem((o) =>
>>>             {
>>>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>>>                 EventConsumer con = new EventConsumer("DTMF", "");
>>>                 while (true)
>>>                 {
>>>                     Event ev = con.pop(1);
>>>                     if (ev != null)
>>>                     {
>>>                                   ... do what ever needs to be done
>>>                     }
>>>
>>>                 }
>>>             });
>>>
>>> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>>>
>>> I am guessing the DTMF is "DTMF" so play around with that if it does not
>>> work.
>>>
>>> There are probably better ways.
>>>
>>>
>>>
>>> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>>
>>>> What i need to do is listen on a session before, during and after the
>>>> session is bridged with another call. I know built into FS is the option to
>>>> listen for *x where x is 0-9 but i really need more than one digit.
>>>>
>>>> I traced the method that listens for the single digit to the
>>>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>>>> switch_ivr_async.c
>>>>
>>>> mod_managed also has the following in the native class freeswitch.
>>>> public static switch_status_t
>>>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>>>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>>> recv_dtmf)
>>>>
>>>> The problem is i can't work out how to use it.  I am fine the the first
>>>> param for the session but have no idea what i need for
>>>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>>> recv_dtmf param.
>>>>
>>>> if anyone can point me in the right direction or knows a better way i
>>>> would be grateful.
>>>>
>>>> Chris
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Wed Mar 10 13:52:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 16:52:29 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
Message-ID: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>

Chris, Thank you.

I find the following information from wiki.This is say sofia.conf. I think
sofia.conf.xml

Multiple Registrations

Call one extension and ring several phones

You must enable multiple registrations in sofia.conf

<param name="multiple-registrations" value="true"/>


----

Which file I should enable internal.xml or sofia.conf.xml

Thanks

Lloyd




On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Hi Lloyd,
>
>
>
> You need to use sofia_contact in your bridge command in the dialplan:
>
>
>
> e.g.
>
> *<anti-action application="bridge"
> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>
> * *
>
> This will cause FS to try all registrations simultaneously.
>
>
>
> Cheers, Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Tuesday, March 09, 2010 11:53 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
> device ring
>
>
>
> Hi All,
>
>
>
> I enable the multiple registrations in sip_profiles/internal.xml.
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> Then I restart the freeswitch.
>
>
>
> I use Aastra 9133i and eyebeam register for same extension.
>
>
>
> But randomly only one device ring. Do I need to enable any other settings?
>
>
>
>
> Thanks,
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jbrucehopkins at gmail.com  Wed Mar 10 14:04:58 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 22:04:58 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
	<d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
Message-ID: <d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>

Interestingly I *can* call outbound to the WAN a SIP client registered with
FS via Fring.  I wonder if Fring have some additional nat traversal/SBC
malarkey going on as they proxy the calls.

I can't call outbound to an Eyebeam softphone registered directly wth FS
though - although said Eyebeam is registered and can call inwards.

I must have my nat handling configured wrongly.

B

On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hey Roly,
>
> Are you getting anywhere?  I'm still struggling, I'm afraid.
>
> I'm able to call from WAN --> LAN using the internal profile on port 5060,
> but I still can't call out from LAN --> WAN so there is obviously something
> really basic I'm still not getting.
>
> In sip_profiles/internal.xml I have:
>
>
>  <param name="aggressive-nat-detection" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
> <!-- the external_rtp_ip etc being defined in vars.xml -->
>
>
> In this example I have left <param name="force-register-domain"
> value="$${domain}"/> etc uncommented to force the registration to go to
> default.xml in the dialplan.
>
> I thought maybe this was my problem and tried putting a separate file in
> the directory to enable the WAN phone to register using the external DNS
> name for the server as its domain, and uncommenting force-register-domain,
> etc.  But then I got an error saying that the local (to Freeswitch) phone in
> the LAN was not registered when I tried to make the call from the WAN.
>
> Any thoughts?
>
> cheers
> Bruce
>
>
>
>
>
>
>
> On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi Roly,
>>
>> I hope to be able to have a go at this later today.  I'll let you know
>> what happens.
>>
>> I have my FS box behind a hardware router/firewall, so I'm assuming I just
>> forward SIP port 5060 from the WAN to the FS box and use internal profile,
>> instead of using external profile with port 5080 forwarded as I have at the
>> moment.  We'll see what happens.
>>
>> cheers
>> Bruce
>>
>> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>>
>>> Ah I see... what? Please share and lead this blind man out the FS
>>> wilderness!
>>>
>>> I don't understand...what happens to the external profile? Do you delete
>>> it?
>>> And how do you forward port 5060?
>>>
>>> ...and you thought you were a newbie!
>>>
>>> Any insight would be much appreciated...loving the journey.
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Bruce
>>> Hopkins
>>> Sent: 09 March 2010 08:23 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>>
>>> Ah I see.  I will try again using the internal profile and forwarding
>>> port 5060.  Presumably still creating a directory entry to enable the
>>> outside-facing domain to be used.
>>>
>>> Many thanks for your patient help of a newbie Brian.
>>>
>>> Cheers
>>> Bruce
>>>
>>> Please excuse the brevity - sent from my mobile.
>>>
>>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>>
>>> > ok you aren't catching one key thing here.. you no longer need two
>>> > profiles.
>>> >
>>> > /b
>>> >
>>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>>> >
>>> >> Hi again,
>>> >>
>>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>>> >> something wrong here as I am still having difficulty, and still
>>> >> have the same problem.
>>> >>
>>> >> I updated to build 16938 by means of "make current"
>>> >>
>>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>>> >> to use 5080 instead of 5090.
>>> >>
>>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>>> >> the server as a domain for the remote softphone to register on.  I
>>> >> left the group name in this entry the same as inthe default entry,
>>> >> so that the remote phone could register on the same extension
>>> >> numbers (100, etc) as in the default build.
>>> >>
>>> >> I still find that, if I initiate a call from the local (on same LAN
>>> >> as freeswitch) phone to the remote phone, I get the message on the
>>> >> CLI:
>>> >>
>>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>>> >> type [user] cause: [USER_NOT_REGISTERED]
>>> >>
>>> >> One possibly unrelated aside, I also found I needed to uncomment
>>> >> <param name="aggressive-nat-detection" value="true"/> in
>>> >> external.xml, otherwise in the case of a call initiated by the
>>> >> remote phone being hung up by the local phone, freeswitch sent the
>>> >> BYE to the private IP of the remote phone, rather than its public
>>> >> ip - meaning that the remote phone didn't receive the BYE.
>>> >>
>>> >> Any further ideas where I am going wrong here please?
>>> >>
>>> >> thanks again in advance
>>> >> Bruce
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From gmaruzz at celliax.org  Wed Mar 10 14:20:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 10 Mar 2010 23:20:04 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B980E77.8080303@meanie.us>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com> 
	<4B8569CB.6070804@meanie.us>
	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com> 
	<4B980E77.8080303@meanie.us>
Message-ID: <7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>

On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre at meanie.us> wrote:

> With this update coming, I am pretty excited to be trying it out. ?I
> have been trying very, very hard to be patient... :-D

Ciao Tim

hehehe, you've missed the announce, so!

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.html




-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From chris.chen2004 at gmail.com  Wed Mar 10 14:21:51 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 10 Mar 2010 17:21:51 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <507898381003101421j493a075cu7e07b14bd13133ae@mail.gmail.com>

Lloyd,
Just the one in internal.xml should do the work.


On Wed, Mar 10, 2010 at 4:52 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Chris, Thank you.
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
> Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
> <param name="multiple-registrations" value="true"/>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>
> Thanks
>
> Lloyd
>
>
>
>
> On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
>
>>  Hi Lloyd,
>>
>>
>>
>> You need to use sofia_contact in your bridge command in the dialplan:
>>
>>
>>
>> e.g.
>>
>> *<anti-action application="bridge"
>> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>>
>> * *
>>
>> This will cause FS to try all registrations simultaneously.
>>
>>
>>
>> Cheers, Chris.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
>> Lloyd
>> *Sent:* Tuesday, March 09, 2010 11:53 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
>> device ring
>>
>>
>>
>> Hi All,
>>
>>
>>
>> I enable the multiple registrations in sip_profiles/internal.xml.
>>
>>
>>
>> <param name="multiple-registrations" value="true"/>
>>
>>
>>
>> Then I restart the freeswitch.
>>
>>
>>
>> I use Aastra 9133i and eyebeam register for same extension.
>>
>>
>>
>> But randomly only one device ring. Do I need to enable any other settings?
>>
>>
>>
>>
>> Thanks,
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 10 14:23:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Mar 2010 14:23:01 -0800
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <87f2f3b91003101423n740fd293p227550ceaca82cfb@mail.gmail.com>

On Wed, Mar 10, 2010 at 1:52 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Chris, Thank you.
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
> Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
> <param name="multiple-registrations" value="true"/>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>

It appears commented out in internal.xml so that's a pretty good hint of
which file it goes in...
-MC
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From gmaruzz at celliax.org  Wed Mar 10 14:23:45 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 10 Mar 2010 23:23:45 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <021b01cac093$c76d3930$5647ab90$@com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com> 
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com> 
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com> 
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com> 
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com> 
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com> 
	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com> 
	<021b01cac093$c76d3930$5647ab90$@com>
Message-ID: <7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>

On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
> OK, can you decode what this means exactly?? I?d be happy to write something
> up to reference users to, but I don?t really know what it means.? It is a a
> quad-core 2.8GHz running CentOS 5.3 64-bit.? It looks like it calculated
> 150, but is that good, bad, indifferent?? I?ve tried to google it, but
> without really knowing what it is I am looking for, I am not getting good
> results.

This tells you the calculated average discrepancy of the timing is 150
microseconds (millisecond/1000), that I believe is good. (hope this
explanation is true ;) )


>
>
>
>
>
> 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating timer,
> please wait...
>
> 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2001 Step: 50
>
> 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
> 1997 Step: 50
>
> 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 1038 Step: 50
>
> 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average: 1018
> Step: 50
>
> 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2018 Step: 50
>
> 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
> 1578 Step: 50
>
> 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of 150
> calculated
>
> 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [CORE_SOFTTIMER_MODULE]
>
> 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
> Timer 'soft'
>
> 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [CORE_PCM_MODULE]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From chris at fowler.cc  Wed Mar 10 14:26:49 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 10 Mar 2010 17:26:49 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only
	one	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>

Hi Lloyd,

It depends on which profile your phones are registering with.  I suspect internal.xml will work.

Chris.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius Lloyd
Sent: Wednesday, March 10, 2010 1:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Chris, Thank you.

I find the following information from wiki.This is say sofia.conf. I think sofia.conf.xml

Multiple Registrations

Call one extension and ring several phones

You must enable multiple registrations in sofia.conf



<param name="multiple-registrations" value="true"/>



----

Which file I should enable internal.xml or sofia.conf.xml

Thanks

Lloyd



On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
Hi Lloyd,

You need to use sofia_contact in your bridge command in the dialplan:

e.g.
<anti-action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

This will cause FS to try all registrations simultaneously.

Cheers, Chris.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Aloysius Lloyd
Sent: Tuesday, March 09, 2010 11:53 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd

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

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From anthony.minessale at gmail.com  Wed Mar 10 14:31:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 16:31:24 -0600
Subject: [Freeswitch-users] sofia_contact strange behavior
In-Reply-To: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>
References: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>
Message-ID: <191c3a031003101431u4ff81841nc19115c8f0bbaf46@mail.gmail.com>

if you pulled that from the user's xml you need to replace dialed_ext with
something else.
That is a special var only relevant as a user param.


On Wed, Mar 10, 2010 at 2:52 PM, RobertT <siniypin at gmail.com> wrote:

> Hello everybody!
>
> I am experiencing problems with sofia_contact command.
> When I dial sofia_contact external/1000@<domain> in fs cli I see correct
> registration data, provided the user 1000 is registered.
> But in the same time when I call ${sofia_conatct(external/${dialed_ext}@$${domain})}
> from dialplan as an argument to deflect application I recieve
> error\user_not_registered.
> What the hell?
>
> Best regards, RobertT.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ledoktre at meanie.us  Wed Mar 10 14:42:21 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 10 Mar 2010 16:42:21 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
	<4B8569CB.6070804@meanie.us>	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
	<4B980E77.8080303@meanie.us>
	<7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>
Message-ID: <4B98204C.2080601@meanie.us>

Giovanni,

Thanks for your reply.  I wondered if I somehow missed it !!

Giovanni Maruzzelli wrote:
> On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre at meanie.us> wrote:
>
> Ciao Tim
>
> hehehe, you've missed the announce, so!
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.html
>   


From jbrucehopkins at gmail.com  Wed Mar 10 15:05:00 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 23:05:00 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
	<d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
	<d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>
Message-ID: <d03a10e01003101505w116f3a38pfe09a29ffdf498b6@mail.gmail.com>

OK - I have put a Wireshark trace across the router between FS and the
internet, and it will not surprise you to hear that when I tried ot call
from LAN-->WAN, FS was sending packets to the private IP of the remote phone
(behind its own NAT).

So how to sort this?

I tried:

1. Commenting these out in the internal profile

<!--all inbound reg will look in this domain for the users -->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--force the domain in subscriptions to this value -->
    <!--<param name="force-subscription-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->

2. then creating a n antry in the directory refering to the same default
group of extensions as the default.xml, but with:
<domain name="dns.aname.for.FS">

and in the variables:

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

which I hoped from what I read in the Wiki might do the trick vis a vis
forceing FS to send the SIP packets to the public IP of the remote phone.

Alas it does not seem to work, and even phoning in from the WAS -->LAN now I
see USER_NOT_REGISTERED errors .  I get the same error in both directions.

Please please please could somebody put me out of my (your ;-) ) misery here
....?

What do I configure to tell FS to send outbound INVITE's to the public IP
the remote phone registered from behind, rather than its private NAT'ed ip?

many thanks in advance.
Bruce




On 10 March 2010 22:04, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Interestingly I *can* call outbound to the WAN a SIP client registered
> with FS via Fring.  I wonder if Fring have some additional nat traversal/SBC
> malarkey going on as they proxy the calls.
>
> I can't call outbound to an Eyebeam softphone registered directly wth FS
> though - although said Eyebeam is registered and can call inwards.
>
> I must have my nat handling configured wrongly.
>
> B
>
>
> On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hey Roly,
>>
>> Are you getting anywhere?  I'm still struggling, I'm afraid.
>>
>> I'm able to call from WAN --> LAN using the internal profile on port 5060,
>> but I still can't call out from LAN --> WAN so there is obviously something
>> really basic I'm still not getting.
>>
>> In sip_profiles/internal.xml I have:
>>
>>
>>  <param name="aggressive-nat-detection" value="true"/>
>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>
>> <!-- the external_rtp_ip etc being defined in vars.xml -->
>>
>>
>> In this example I have left <param name="force-register-domain"
>> value="$${domain}"/> etc uncommented to force the registration to go to
>> default.xml in the dialplan.
>>
>> I thought maybe this was my problem and tried putting a separate file in
>> the directory to enable the WAN phone to register using the external DNS
>> name for the server as its domain, and uncommenting force-register-domain,
>> etc.  But then I got an error saying that the local (to Freeswitch) phone in
>> the LAN was not registered when I tried to make the call from the WAN.
>>
>> Any thoughts?
>>
>> cheers
>> Bruce
>>
>>
>>
>>
>>
>>
>>
>> On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>>
>>> Hi Roly,
>>>
>>> I hope to be able to have a go at this later today.  I'll let you know
>>> what happens.
>>>
>>> I have my FS box behind a hardware router/firewall, so I'm assuming I
>>> just forward SIP port 5060 from the WAN to the FS box and use internal
>>> profile, instead of using external profile with port 5080 forwarded as I
>>> have at the moment.  We'll see what happens.
>>>
>>> cheers
>>> Bruce
>>>
>>> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>>>
>>>> Ah I see... what? Please share and lead this blind man out the FS
>>>> wilderness!
>>>>
>>>> I don't understand...what happens to the external profile? Do you delete
>>>> it?
>>>> And how do you forward port 5060?
>>>>
>>>> ...and you thought you were a newbie!
>>>>
>>>> Any insight would be much appreciated...loving the journey.
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>>> Bruce
>>>> Hopkins
>>>> Sent: 09 March 2010 08:23 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>>>
>>>> Ah I see.  I will try again using the internal profile and forwarding
>>>> port 5060.  Presumably still creating a directory entry to enable the
>>>> outside-facing domain to be used.
>>>>
>>>> Many thanks for your patient help of a newbie Brian.
>>>>
>>>> Cheers
>>>> Bruce
>>>>
>>>> Please excuse the brevity - sent from my mobile.
>>>>
>>>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> > ok you aren't catching one key thing here.. you no longer need two
>>>> > profiles.
>>>> >
>>>> > /b
>>>> >
>>>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>>>> >
>>>> >> Hi again,
>>>> >>
>>>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>>>> >> something wrong here as I am still having difficulty, and still
>>>> >> have the same problem.
>>>> >>
>>>> >> I updated to build 16938 by means of "make current"
>>>> >>
>>>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>>>> >> to use 5080 instead of 5090.
>>>> >>
>>>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>>>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>>>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>>>> >> the server as a domain for the remote softphone to register on.  I
>>>> >> left the group name in this entry the same as inthe default entry,
>>>> >> so that the remote phone could register on the same extension
>>>> >> numbers (100, etc) as in the default build.
>>>> >>
>>>> >> I still find that, if I initiate a call from the local (on same LAN
>>>> >> as freeswitch) phone to the remote phone, I get the message on the
>>>> >> CLI:
>>>> >>
>>>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>>>> >> type [user] cause: [USER_NOT_REGISTERED]
>>>> >>
>>>> >> One possibly unrelated aside, I also found I needed to uncomment
>>>> >> <param name="aggressive-nat-detection" value="true"/> in
>>>> >> external.xml, otherwise in the case of a call initiated by the
>>>> >> remote phone being hung up by the local phone, freeswitch sent the
>>>> >> BYE to the private IP of the remote phone, rather than its public
>>>> >> ip - meaning that the remote phone didn't receive the BYE.
>>>> >>
>>>> >> Any further ideas where I am going wrong here please?
>>>> >>
>>>> >> thanks again in advance
>>>> >> Bruce
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> > users
>>>> > http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
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From anthony.minessale at gmail.com  Wed Mar 10 15:12:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 17:12:44 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
	<021b01cac093$c76d3930$5647ab90$@com>
	<7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>
Message-ID: <191c3a031003101512u77744760h375c94f05d42a9fa@mail.gmail.com>

it means that it has detected a 150 microsecond offset.

It tries originally to sleep 1000 microseconds (1 millisecond) in a loop.
It times how long the sleep lasted and compares how long it really took vs
how long it was asked to sleep.
then it slowly decreases by 50 and tries again until it gets within 2
microseconds of the goal.

in your case of 150, it means if you subtract 150 from the total number of
microseconds from all the sleeps
they will be more accurate.



On Wed, Mar 10, 2010 at 4:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
> > OK, can you decode what this means exactly?  I?d be happy to write
> something
> > up to reference users to, but I don?t really know what it means.  It is a
> a
> > quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
> > 150, but is that good, bad, indifferent?  I?ve tried to google it, but
> > without really knowing what it is I am looking for, I am not getting good
> > results.
>
> This tells you the calculated average discrepancy of the timing is 150
> microseconds (millisecond/1000), that I believe is good. (hope this
> explanation is true ;) )
>
>
> >
> >
> >
> >
> >
> > 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating
> timer,
> > please wait...
> >
> > 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000
> Average:
> > 2001 Step: 50
> >
> > 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
> > 1997 Step: 50
> >
> > 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 1038 Step: 50
> >
> > 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 1018
> > Step: 50
> >
> > 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000
> Average:
> > 2018 Step: 50
> >
> > 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
> > 1578 Step: 50
> >
> > 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of
> 150
> > calculated
> >
> > 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [CORE_SOFTTIMER_MODULE]
> >
> > 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
> > Timer 'soft'
> >
> > 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [CORE_PCM_MODULE]
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 10 15:15:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 17:15:59 -0600
Subject: [Freeswitch-users] Tone Detection During Bridge
In-Reply-To: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>
References: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>
Message-ID: <191c3a031003101515r415299dckdea8d73369c0d89b@mail.gmail.com>

it should be running during the bridge as well.
once you start it, it attaches in the background and snoops all of your
audio.

Try turning up the debug level to see if it may be exiting.

"console loglevel debug"


On Wed, Mar 10, 2010 at 3:05 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Dear List,
>
> I have a requirement (due to poor quality Sipura products) to have
> freeswitch listen for a tone during a bridge (standard local extension
> dialplan segment). I have the required line:
>
> <action appication="tone_detect" data="hangup 400 r 0 hangup
> 'normal_clearing' 1"/>
>
> But I need to have this run _during_ the bridge (to listen for the tone
> coming from the broken SPA3k and to break the bridge). Does anyone have any
> idea how this may be achieved? There is no such documentation on the wiki.
> This works perfectly if the bridge was not successful, and so will detect
> tones during voicemail messaging.
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ben at langfeld.co.uk  Wed Mar 10 15:57:02 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 10 Mar 2010 23:57:02 +0000
Subject: [Freeswitch-users] Tone Detection During Bridge (Anthony
	Minessale)
Message-ID: <fb4ab1e41003101557r733f148o72f87276f11d87b3@mail.gmail.com>

Cheers Anthony, I don't know how I was overcomplicating that so much, but it
works fine now. Thanks!

Regards,
Ben Langfeld

Wave > Email


On Wed, Mar 10, 2010 at 11:16 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
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>
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> or, via email, send a message with subject or body 'help' to
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> When replying, please edit your Subject line so it is more specific
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>
> Today's Topics:
>
>   1. Re: Kernel timer warning with CentOS 5.4 (Anthony Minessale)
>   2. Re: Tone Detection During Bridge (Anthony Minessale)
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 10 Mar 2010 17:12:44 -0600
> Subject: Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4
> it means that it has detected a 150 microsecond offset.
>
> It tries originally to sleep 1000 microseconds (1 millisecond) in a loop.
> It times how long the sleep lasted and compares how long it really took vs
> how long it was asked to sleep.
> then it slowly decreases by 50 and tries again until it gets within 2
> microseconds of the goal.
>
> in your case of 150, it means if you subtract 150 from the total number of
> microseconds from all the sleeps
> they will be more accurate.
>
>
>
> On Wed, Mar 10, 2010 at 4:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:
>
>> On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
>> > OK, can you decode what this means exactly?  I?d be happy to write
>> something
>> > up to reference users to, but I don?t really know what it means.  It is
>> a a
>> > quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
>> > 150, but is that good, bad, indifferent?  I?ve tried to google it, but
>> > without really knowing what it is I am looking for, I am not getting
>> good
>> > results.
>>
>> This tells you the calculated average discrepancy of the timing is 150
>> microseconds (millisecond/1000), that I believe is good. (hope this
>> explanation is true ;) )
>>
>>
>> >
>> >
>> >
>> >
>> >
>> > 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating
>> timer,
>> > please wait...
>> >
>> > 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000
>> Average:
>> > 2001 Step: 50
>> >
>> > 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650
>> Average:
>> > 1997 Step: 50
>> >
>> > 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 1038 Step: 50
>> >
>> > 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 1018
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000
>> Average:
>> > 2018 Step: 50
>> >
>> > 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900
>> Average:
>> > 1578 Step: 50
>> >
>> > 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of
>> 150
>> > calculated
>> >
>> > 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
>> > Successfully Loaded [CORE_SOFTTIMER_MODULE]
>> >
>> > 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
>> > Timer 'soft'
>> >
>> > 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
>> > Successfully Loaded [CORE_PCM_MODULE]
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 10 Mar 2010 17:15:59 -0600
> Subject: Re: [Freeswitch-users] Tone Detection During Bridge
> it should be running during the bridge as well.
> once you start it, it attaches in the background and snoops all of your
> audio.
>
> Try turning up the debug level to see if it may be exiting.
>
> "console loglevel debug"
>
>
> On Wed, Mar 10, 2010 at 3:05 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Dear List,
>>
>> I have a requirement (due to poor quality Sipura products) to have
>> freeswitch listen for a tone during a bridge (standard local extension
>> dialplan segment). I have the required line:
>>
>> <action appication="tone_detect" data="hangup 400 r 0 hangup
>> 'normal_clearing' 1"/>
>>
>> But I need to have this run _during_ the bridge (to listen for the tone
>> coming from the broken SPA3k and to break the bridge). Does anyone have any
>> idea how this may be achieved? There is no such documentation on the wiki.
>> This works perfectly if the bridge was not successful, and so will detect
>> tones during voicemail messaging.
>>
>> Regards,
>> Ben Langfeld
>>
>> Wave > Email
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Wed Mar 10 16:41:55 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 11 Mar 2010 03:41:55 +0300
Subject: [Freeswitch-users] Switch with External SoftPhone
Message-ID: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>

Hello.

What are the ways connect external users to the fs there? In such a scheme:
 fs with real ip <-> internet <-> nat <-> sip-phone


I read the topic http://wiki.freeswitch.org/wiki/External_profile #
Switch_with_External_SoftPhone

I did not understand several things:
- That should be in exteranl5090.xml? Whole profile external.xml c changes
in paragraphs 2,3?
- Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in
xxxx? External UA IP? Domain?" I have the same question. that x.x.x.x should
be?

But this seems not quite what I need. Must be external to the subscriber is
practically no different from internal.

-- 
Best regards, Vladimir Elizarov
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From lloyd.aloysius at gmail.com  Wed Mar 10 16:59:06 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 19:59:06 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>
Message-ID: <8a19bf2e1003101659x29ba2620k888b4a4fc2a3b546@mail.gmail.com>

Thank you for the help.



On Wed, Mar 10, 2010 at 5:26 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Hi Lloyd,
>
>
>
> It depends on which profile your phones are registering with.  I suspect
> internal.xml will work.
>
>
>
> Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Wednesday, March 10, 2010 1:52 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Multiple Registrations - randomly only
> one device ring
>
>
>
> Chris, Thank you.
>
>
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
>
>  Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>
> Thanks
>
> Lloyd
>
>
>
>
>
> On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
>
> Hi Lloyd,
>
>
>
> You need to use sofia_contact in your bridge command in the dialplan:
>
>
>
> e.g.
>
> *<anti-action application="bridge"
> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>
> * *
>
> This will cause FS to try all registrations simultaneously.
>
>
>
> Cheers, Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Tuesday, March 09, 2010 11:53 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
> device ring
>
>
>
> Hi All,
>
>
>
> I enable the multiple registrations in sip_profiles/internal.xml.
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> Then I restart the freeswitch.
>
>
>
> I use Aastra 9133i and eyebeam register for same extension.
>
>
>
> But randomly only one device ring. Do I need to enable any other settings?
>
>
>
>
> Thanks,
> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Wed Mar 10 20:07:06 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 11 Mar 2010 09:37:06 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
	<7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
	<F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>
Message-ID: <7d79b3931003102007n728e2c3dudf603aafedb18b6b@mail.gmail.com>

My switch type is EuroISDN, and I used Q931 as the dialect.


On Wed, Mar 10, 2010 at 3:31 PM, Michael Jerris <mike at jerris.com> wrote:

> What pri dialect is this?
>
> On Mar 4, 2010, at 11:11 PM, lakshmanan ganapathy wrote:
>
> Ok. This is to say how the problem got solved.
> Need openzap 1047 or above version.
> Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz
> My telco is not accepting the display IE.
> Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.
>
> Thanks for all.
>
> On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn <tomek.augustyn at gmail.com
> > wrote:
>
>> Hello,
>>
>> I had similar problem and I think it is more a problem between Sangoma
>> card and your E1 provider than with freeswitch.
>>
>> In my case it was necessary to set "origination_caller_id_number" to one
>> of the telephone numbers linked to my E1 line. In other case the calls were
>> rejected with "invalid information element" error.
>>
>> You can try Sangoma's support they are very helpful.
>>
>> Tomasz Augustyn
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lvatland at gmail.com  Wed Mar 10 07:31:28 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Wed, 10 Mar 2010 09:31:28 -0600
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
Message-ID: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?


Thanks for any help.


Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com



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From lvatland at gmail.com  Wed Mar 10 14:38:19 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Wed, 10 Mar 2010 16:38:19 -0600
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
Message-ID: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?


Thanks for any help.


Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com

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From arliyardo at gmail.com  Wed Mar 10 17:25:50 2010
From: arliyardo at gmail.com (arliyardo)
Date: Thu, 11 Mar 2010 09:25:50 +0800
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls between two NAT?
Message-ID: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>

my scenario like this:

PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET ->
NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
First, I have forwarded sip port and rtp port.
I have try servaral configurations in
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and
without luck!

PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2,
include sip signaling and rtp connection ?
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From helmut.kuper at ewetel.de  Thu Mar 11 00:34:41 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 11 Mar 2010 09:34:41 +0100
Subject: [Freeswitch-users] openzap  Q.SIG and Q.931
Message-ID: <4B98AB21.5010102@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

maybe you know that I run a long time test with Stefan Knoblich's q931
pri-stack for openzap. In fact, I use is in my production system since a
year or so. I have to say that it is very stable and I'm happy with it -
ususally ... but:

I wonder if anyone of you is working on a Q.SIG Basic-Call Support
(ECMA-143) for openzap. I try to connect to an Avaya system using Q.SIG
to be a part of the internal domain. My main targets are to send and
receive Caller's and Callee's name and to remove a leading '0' (which is
added by AVAYA because of q931) for internal calls.

regards
helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLmKsh4tZeNddg3dwRAjaNAJ4pTdxYyQizHPDXP9gjjUyABJIuxwCcDnNj
qlCG9cdIfjDLlQE3bQfxluw=
=L1tJ
-----END PGP SIGNATURE-----


From mike at jerris.com  Thu Mar 11 01:09:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Mar 2010 04:09:03 -0500
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
In-Reply-To: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
References: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
Message-ID: <DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>

I think I fixed this about a week ago.

On Mar 10, 2010, at 10:31 AM, Larry Vatland wrote:

> I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?
> 



From jbrucehopkins at gmail.com  Thu Mar 11 02:03:39 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Thu, 11 Mar 2010 10:03:39 +0000
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls 	between two NAT?
In-Reply-To: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
References: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
Message-ID: <d03a10e01003110203t29ebc270l73594f28463b1e96@mail.gmail.com>

Hello arliyardo,

I'm new to Freeswitch, but looking at the posts on the mailing list it
appears a few of us - you, me and Vladimir Elizarov, have been grappling
with he same thing recently - getting remote phones to work through NAT.

If you look at my thread you will see I have got some of the way, and can
call incoming now but still have difficulty with outgoing calls.  Perhaps
you might have a look at where I have got to and see if you can think of the
final bit of the picture?

I gather the developers are in the process of updating the Wiki pages on
NAT.  I'm hoping they will be able to find time to continuw with this.  It
must be a laborious and time consuming task keeping the documentation up to
date.

bests
Bruce

On 11 March 2010 01:25, arliyardo <arliyardo at gmail.com> wrote:

> my scenario like this:
>
> PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET -> NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
> First, I have forwarded sip port and rtp port.
> I have try servaral configurations in http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and without luck!
>
> PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2, include sip signaling and rtp connection ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Mar 11 06:10:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Mar 2010 08:10:37 -0600
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
In-Reply-To: <DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>
References: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
	<DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>
Message-ID: <6A9394D0-14C4-489C-8018-B79D1E741A8F@freeswitch.org>

I don't think... I know you fixed it about a week ago :P

/b

On Mar 11, 2010, at 3:09 AM, Michael Jerris wrote:

> I think I fixed this about a week ago.



From phunk0000 at hotmail.com  Thu Mar 11 07:44:02 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 11 Mar 2010 10:44:02 -0500
Subject: [Freeswitch-users] can't get programming language modules to load
Message-ID: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>

Hey list-  I have make mod_whatever-installed all the individual mod I want
and have been able to make all but the following work.  I have followed wiki
directions for odbc and have odbc working.  Any help in getting these
modules to load would be great.  Thanks

 

**/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
mod_spidermonkey_core_db_module_interface**

2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so

**/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
mod_spidermonkey_curl_module_interface**

2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so

**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
mod_spidermonkey_odbc_module_interface**

2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so

**/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
mod_spidermonkey_socket_module_interface**

2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so

**/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
mod_spidermonkey_teletone_module_interface**

2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_perl.so

**/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
such file or directory**

2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_python.so

**/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
No such file or directory**

2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so

**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such file or directory**

 

Thanks

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From brian at freeswitch.org  Thu Mar 11 07:51:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Mar 2010 09:51:06 -0600
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
Message-ID: <42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>

Those are sub modules of spidermonkey not FreeSWITCH.... Please see the spidermonkey.conf.xml

And the others are not compiled.

/b

On Mar 11, 2010, at 9:44 AM, Todd wrote:

> Hey list-  I have make mod_whatever-installed all the individual mod I want and have been able to make all but the following work.  I have followed wiki directions for odbc and have odbc working.  Any help in getting these modules to load would be great.  Thanks
>  
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol: mod_spidermonkey_core_db_module_interface**
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface**
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: mod_spidermonkey_odbc_module_interface**
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol: mod_spidermonkey_socket_module_interface**
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol: mod_spidermonkey_teletone_module_interface**
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_perl.so
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No such file or directory**
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_python.so
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: No such file or directory**
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No such file or directory**
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From phunk0000 at hotmail.com  Thu Mar 11 08:31:24 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 11 Mar 2010 11:31:24 -0500
Subject: [Freeswitch-users] can't get programming language modules
	to	load
In-Reply-To: <42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
Message-ID: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>

Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
still get those same errors.  I have also attempted to 'make
mod_java-install' for all the uncompiled ones and it runs through a bunch of
stuff on the screen but ends with 'make[1]: *** [mod_java.lo] error 1' same
with python and lua.  any info on how to properly compile the others and get
the spidermokney subs to work would be great.  thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 11, 2010 10:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] can't get programming language modules to
load

 

Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
spidermonkey.conf.xml

 

And the others are not compiled.

 

/b

 

On Mar 11, 2010, at 9:44 AM, Todd wrote:





Hey list-  I have make mod_whatever-installed all the individual mod I want
and have been able to make all but the following work.  I have followed wiki
directions for odbc and have odbc working.  Any help in getting these
modules to load would be great.  Thanks

 

**/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
mod_spidermonkey_core_db_module_interface**

2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so

**/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
mod_spidermonkey_curl_module_interface**

2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so

**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
mod_spidermonkey_odbc_module_interface**

2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so

**/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
mod_spidermonkey_socket_module_interface**

2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so

**/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
mod_spidermonkey_teletone_module_interface**

2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_perl.so

**/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
such file or directory**

2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_python.so

**/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
No such file or directory**

2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so

**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such file or directory**

 

Thanks

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

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
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From tculjaga at gmail.com  Thu Mar 11 09:15:16 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Mar 2010 18:15:16 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
	<191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
Message-ID: <65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>

On Wed, Mar 10, 2010 at 6:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We have a mod_fax for t30 and work on t38 is underway.
> So we already acknowledge that I guess.
>
>
>
anything ready for testing ?
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From tculjaga at gmail.com  Thu Mar 11 09:19:01 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Mar 2010 18:19:01 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <65d96fc81003110919i7a9bcc87xd92edeab6598fc0b@mail.gmail.com>

On Wed, Mar 10, 2010 at 5:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>
Right, this happens on 32bit Centos (at least in my case) only... on 64bit
is is ok ... and no complains at all :)

misery solved for me.... thx.


T.
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From ken at ukgb.net  Thu Mar 11 09:34:08 2010
From: ken at ukgb.net (Ken Gillett)
Date: Thu, 11 Mar 2010 17:34:08 +0000
Subject: [Freeswitch-users] PSTN connection
Message-ID: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>

If I am running FreeSwitch on a Mac, would I be able to use Apple's USB modem as a means to connect to the PSTN and utilise this line to place and receive calls?



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From freeswitch at cartissolutions.com  Thu Mar 11 10:35:30 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 11 Mar 2010 12:35:30 -0600
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>	<5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
	<8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>
Message-ID: <4B9937F2.2010807@cartissolutions.com>


On 03/10/2010 04:12 AM, Dome Charoenyost wrote:
> 2010/3/10 Michael Jerris<mike at jerris.com>:
>    
>> -nf has nothing to do with the performance, it has to do if the command
>> blocks or not when you run it.  I am unsure what you mean about sql in this
>> context, it is totally unrelated. What is a cash db?
>>      
> sorry i mean switch_cache_db_connection :)
> i have problem about db connection. now switch_cache_db_connection use
> 1 connection for 1 thread
> if i use mod_limit with odbc my server can't handle more than 100 connection.
> So i'll try FS in no fork. but i want to know other effect.
>
> BG
>
> Dome C.
>    

I think you misunderstand what it is that the -nf option does.  It will 
have no bearing on how switch's odbc caching mechanisms work.  For 
example, I run freeswitch under a supervising daemon (djb's daemontools 
in this case) with -nf and -nc options.  This prevents freeswitch from 
spawning the console, and it also prevents freeswitch from forking to 
the background.  In otherwords, it keeps freeswitch from becoming a 
background daemon.


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com




From testeador01 at gmail.com  Thu Mar 11 11:24:25 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 11 Mar 2010 14:24:25 -0500
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
	<blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
Message-ID: <c29b930b1003111124m10b85e08t9fae1067a54b4d9b@mail.gmail.com>

Hello Todd,

I know this can be a little complicated at first but if you research the
wiki and google you can succeed,

You didn't need to just "uncomment" them, you need to install them so they
can actually run...

to install spidermonkey odbc, there are actually some packages that have to
be installed on your linux box, you can find them on the wiki for SPIDERMONKEY
ODBC <http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc> not just the
one for odbc, in that wiki, they tell you what you need to install.

to install mod java you need to run the configure option that is indicated
in the mod_java wiki as well...

if you still get errors after that, please post back, but post the whole
error message, saying "error 1" doesn't help us understand what are you
missing.

Good luck with this :)


2010/3/11 Todd <phunk0000 at hotmail.com>

>  Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
> still get those same errors.  I have also attempted to ?make
> mod_java-install? for all the uncompiled ones and it runs through a bunch of
> stuff on the screen but ends with ?make[1]: *** [mod_java.lo] error 1? same
> with python and lua?  any info on how to properly compile the others and get
> the spidermokney subs to work would be great.  thanks
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Thursday, March 11, 2010 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] can't get programming language modules
> to load
>
>
>
> Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
> spidermonkey.conf.xml
>
>
>
> And the others are not compiled.
>
>
>
> /b
>
>
>
> On Mar 11, 2010, at 9:44 AM, Todd wrote:
>
>
>
>   Hey list-  I have make mod_whatever-installed all the individual mod I
> want and have been able to make all but the following work.  I have followed
> wiki directions for odbc and have odbc working.  Any help in getting these
> modules to load would be great.  Thanks
>
>
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
> mod_spidermonkey_core_db_module_interface**
>
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
> mod_spidermonkey_curl_module_interface**
>
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
> mod_spidermonkey_odbc_module_interface**
>
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
> mod_spidermonkey_socket_module_interface**
>
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
> mod_spidermonkey_teletone_module_interface**
>
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_perl.so
>
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
> such file or directory**
>
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_python.so
>
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
> No such file or directory**
>
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so
>
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such file or directory**
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
> 02:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testeador01 at gmail.com  Thu Mar 11 11:24:25 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 11 Mar 2010 14:24:25 -0500
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
	<blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
Message-ID: <c29b930b1003111124m10b85e08t9fae1067a54b4d9b@mail.gmail.com>

Hello Todd,

I know this can be a little complicated at first but if you research the
wiki and google you can succeed,

You didn't need to just "uncomment" them, you need to install them so they
can actually run...

to install spidermonkey odbc, there are actually some packages that have to
be installed on your linux box, you can find them on the wiki for SPIDERMONKEY
ODBC <http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc> not just the
one for odbc, in that wiki, they tell you what you need to install.

to install mod java you need to run the configure option that is indicated
in the mod_java wiki as well...

if you still get errors after that, please post back, but post the whole
error message, saying "error 1" doesn't help us understand what are you
missing.

Good luck with this :)


2010/3/11 Todd <phunk0000 at hotmail.com>

>  Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
> still get those same errors.  I have also attempted to ?make
> mod_java-install? for all the uncompiled ones and it runs through a bunch of
> stuff on the screen but ends with ?make[1]: *** [mod_java.lo] error 1? same
> with python and lua?  any info on how to properly compile the others and get
> the spidermokney subs to work would be great.  thanks
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Thursday, March 11, 2010 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] can't get programming language modules
> to load
>
>
>
> Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
> spidermonkey.conf.xml
>
>
>
> And the others are not compiled.
>
>
>
> /b
>
>
>
> On Mar 11, 2010, at 9:44 AM, Todd wrote:
>
>
>
>   Hey list-  I have make mod_whatever-installed all the individual mod I
> want and have been able to make all but the following work.  I have followed
> wiki directions for odbc and have odbc working.  Any help in getting these
> modules to load would be great.  Thanks
>
>
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
> mod_spidermonkey_core_db_module_interface**
>
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
> mod_spidermonkey_curl_module_interface**
>
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
> mod_spidermonkey_odbc_module_interface**
>
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
> mod_spidermonkey_socket_module_interface**
>
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
> mod_spidermonkey_teletone_module_interface**
>
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_perl.so
>
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
> such file or directory**
>
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_python.so
>
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
> No such file or directory**
>
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so
>
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such file or directory**
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
> 02:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:37:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:37:26 -0800
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
Message-ID: <87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>

How are your internal users registering to your FS? What's your FS setup? Do
you have more than one NIC?
-MC

On Wed, Mar 10, 2010 at 4:41 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Hello.
>
> What are the ways connect external users to the fs there? In such a
> scheme:
>  fs with real ip <-> internet <-> nat <-> sip-phone
>
>
> I read the topic http://wiki.freeswitch.org/wiki/External_profile #
> Switch_with_External_SoftPhone
>
> I did not understand several things:
> - That should be in exteranl5090.xml? Whole profile external.xml c changes
> in paragraphs 2,3?
> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in
> xxxx? External UA IP? Domain?" I have the same question. that x.x.x.x
> should be?
>
> But this seems not quite what I need. Must be external to the subscriber
> is practically no different from internal.
>
> --
> Best regards, Vladimir Elizarov
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:39:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:39:16 -0800
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
Message-ID: <87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>

On Wed, Mar 10, 2010 at 2:38 PM, Larry Vatland <lvatland at gmail.com> wrote:

> I'm new to freeswitch and just exploring the default dialplan. I have not
> been able to get the mad_boss_intercom example to work correctly, it pages
> fine but you should be able to press *2 to start a two-way conversation.
> Whats happening is the * toggles the phones mute. How do I set up the
> conference to pass the *2 off to the bind_meta_app to get a page to intercom
> system working?
>
> What phone are you using? It sounds like maybe the phone itself is
intercepting the * and not sending it along to FreeSWITCH.
-MC
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From msc at freeswitch.org  Thu Mar 11 12:45:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:45:40 -0800
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls 	between two NAT?
In-Reply-To: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
References: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
Message-ID: <87f2f3b91003111245g2b4833a7re68ca87ca1d09353@mail.gmail.com>

On Wed, Mar 10, 2010 at 5:25 PM, arliyardo <arliyardo at gmail.com> wrote:

> my scenario like this:
>
> PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET -> NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
> First, I have forwarded sip port and rtp port.
> I have try servaral configurations in http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and without luck!
>
> PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2, include sip signaling and rtp connection ?
>
>
> The key here is that you have two devices that can totally mess with your
SIP and RTP: NAT1 and NAT2. What are these two devices? Secondly, what kind
of phone is PHONE2? Some phones work better than others. Third, have you
captured the SIP traffic behind each NAT device to see what's going on? You
won't be able to fix the problem until you actually find out what the
problem is and where it is happening. For the record, I've done this kind of
thing using X-Lite and Snom 300's as the PHONE2 in your example, so I know
it is possible. The trick is making sure that your NAT devices have their
SIP ALG's turned off. Check this page to see if you have one of these
devices:
http://wiki.freeswitch.org/wiki/ALG

-MC
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From pjintheusa at gmail.com  Thu Mar 11 12:48:42 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 11 Mar 2010 15:48:42 -0500
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
Message-ID: <367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>

Larry - also see the replies to your previous post on this subject - which I
would paraphrase as "make sure the FS version you are using to test is the
latest"

On Thu, Mar 11, 2010 at 3:39 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Mar 10, 2010 at 2:38 PM, Larry Vatland <lvatland at gmail.com> wrote:
>
>> I'm new to freeswitch and just exploring the default dialplan. I have not
>> been able to get the mad_boss_intercom example to work correctly, it pages
>> fine but you should be able to press *2 to start a two-way conversation.
>> Whats happening is the * toggles the phones mute. How do I set up the
>> conference to pass the *2 off to the bind_meta_app to get a page to intercom
>> system working?
>>
>> What phone are you using? It sounds like maybe the phone itself is
> intercepting the * and not sending it along to FreeSWITCH.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:56:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:56:45 -0800
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
	<367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
Message-ID: <87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>

On Thu, Mar 11, 2010 at 12:48 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Larry - also see the replies to your previous post on this subject - which
> I would paraphrase as "make sure the FS version you are using to test is the
> latest"
>
Or, in other words, go to your FreeSWITCH source directory and type "make
current" and then go get a cup of coffee. When it is done restart FreeSWITCH
and test again...
-MC
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From srinivas.ksvreddy at gmail.com  Thu Mar 11 13:00:54 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 11 Mar 2010 16:00:54 -0500
Subject: [Freeswitch-users] more than one condition default.xml
Message-ID: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>

Hi,

can i have more than one conditions in externsion tag in
dialplan\default.xml.



-- 
Srinivasula Reddy K
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From anthony.minessale at gmail.com  Thu Mar 11 13:09:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Mar 2010 15:09:05 -0600
Subject: [Freeswitch-users] more than one condition default.xml
In-Reply-To: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
References: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
Message-ID: <191c3a031003111309v6d1dcceek41516267e6f3c9f1@mail.gmail.com>

yes but make sure they are stacked not nested (see wiki)

On Thu, Mar 11, 2010 at 3:00 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
> Hi,
>
> can i have more than one conditions in externsion tag in
> dialplan\default.xml.
>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar 11 13:25:18 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 11 Mar 2010 15:25:18 -0600
Subject: [Freeswitch-users] more than one condition default.xml
In-Reply-To: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
References: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
Message-ID: <4B995FBE.4090506@cartissolutions.com>

On 03/11/2010 03:00 PM, srinivasula reddy wrote:
>
> Hi,
> can i have more than one conditions in externsion tag in 
> dialplan\default.xml.
>
> -- 
> Srinivasula Reddy K
>

You can do the following:

<extension name="foo">
<condition ... />
<condition ... />
<condition ... >
<action ... />
<action ... />
<anti-action ... />
</condition>
<condition ... >
<action ... />
<anti-action ... />
</condition>
</extension>

But you can not have nested conditions.


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com




From xengelpublicx at gmail.com  Thu Mar 11 13:25:52 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Fri, 12 Mar 2010 00:25:52 +0300
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
	<87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>
Message-ID: <eb0385731003111325j50686b27t83825ed89856fcfa@mail.gmail.com>

Internal users connects via internal profile.?FS has two interfaces lan and wan.
scheme:
wan(external real ip)<-fs->lan (local ip)-internal users

I need to have external users with the same capabilities as the internal users.

On Thu, Mar 11, 2010 at 11:37 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> How are your internal users registering to your FS? What's your FS setup? Do you have more than one NIC?
> -MC
>
> On Wed, Mar 10, 2010 at 4:41 PM, Vladimir Elizarov <xengelpublicx at gmail.com> wrote:
>>
>> Hello.
>>
>> What are the ways connect external users to the fs there??In such a scheme:
>> ?fs with real ip <-> internet <-> nat <-> sip-phone
>>
>>
>> I read the topic http://wiki.freeswitch.org/wiki/External_profile # Switch_with_External_SoftPhone
>>
>> I did not understand several things:
>> - That should be in exteranl5090.xml??Whole profile external.xml c changes in paragraphs 2,3?
>> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in xxxx? External UA IP? Domain?"?I have the same question.?that x.x.x.x should be?
>>
>> But this seems not quite what I need.?Must be external to the subscriber is practically no different from internal.
>>
>> --
>> Best regards, Vladimir Elizarov
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Best regards, Vladimir Elizarov


From lvatland at gmail.com  Thu Mar 11 13:33:47 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Thu, 11 Mar 2010 15:33:47 -0600
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
	<367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
	<87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>
Message-ID: <FFD74D15-D9F7-408C-AF7A-77C148B0D983@gmail.com>

Thanks All.
The phone isn't the issue, when I push the * I receive a message from freeswitch about mute-on/mute-off. I tried to make current however I get a compile error with the current version see below, this is compiling on Mac OS-X 10.6.2 older version of freeswitch did compile fine.

Compiling /usr/src/freeswitch-snapshot/src/mod/say/mod_say_en/mod_say_en.c ...
Creating mod_say_en.so...
Undefined symbols:
  "_switch_ivr_say_spell", referenced from:
      _en_say in mod_say_en.o
  "_switch_ivr_say_ip", referenced from:
      _en_say in mod_say_en.o
ld: symbol(s) not found
collect2: ld returned 1 exit status


Thanks,
Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com



On Mar 11, 2010, at 2:56 PM, Michael Collins wrote:

> 
> 
> On Thu, Mar 11, 2010 at 12:48 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Larry - also see the replies to your previous post on this subject - which I would paraphrase as "make sure the FS version you are using to test is the latest"
> Or, in other words, go to your FreeSWITCH source directory and type "make current" and then go get a cup of coffee. When it is done restart FreeSWITCH and test again...
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From irmatov at gmail.com  Thu Mar 11 23:20:42 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 12 Mar 2010 12:20:42 +0500
Subject: [Freeswitch-users] RFC3578
Message-ID: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>

Hi,

Is there support for RFC3578 in FreeSWITCH? If not, is there any plans
to implement it?

As far as I understand from quick browsing through this RFC, it is
about mapping of ISUP overlap signalling to SIP protocol. In short,
remote device (softswitch for example) sends short (incomplete) number
in INVITE message to freeswitch, which responds with '484 address
incomplete', then softswitch send another INVITE with more digits.
Freeswitch then can send another 484, and after some time (enough
digits collected), proceed with usual call establishment.

-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From skoost at skoost.com  Thu Mar 11 23:36:31 2010
From: skoost at skoost.com (Thangappan Mohan)
Date: 12 Mar 2010 07:36:31 +0000
Subject: [Freeswitch-users] Thangappan Mohan sent you a little gift
Message-ID: <20100312073631.D5C2D17F741@skoismta01.skoost.com>

Thangappan Mohan is on Skoost and sent you a little gift.

To collect your gift, follow the link below:
http://www.skoost.com/?id=114961367_5284543

P.S. This is a safe and innocent gift that Thangappan Mohan sent from Skoost.

Skoost

From gavin.henry at gmail.com  Fri Mar 12 02:22:02 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Mar 2010 10:22:02 +0000
Subject: [Freeswitch-users] UK ring tone
Message-ID: <13ca621c1003120222k5061879ei98863652c6ebd2db@mail.gmail.com>

Hi,

I've checked the internal and external sip profiles and think this
needs to be set in autoload switch file as per:

http://wiki.freeswitch.org/wiki/Variable_transfer_ringback

I want by default a UK ringtone to be heard when I call a DDI that is
configured on the system. At the moment it sounds like I'm calling the
US.

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From brian at freeswitch.org  Fri Mar 12 04:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Mar 2010 06:56:56 -0600
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
Message-ID: <B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>

It already works use the respond application in the dialplan to respond 484

/b

On Mar 12, 2010, at 1:20 AM, Timur Irmatov wrote:

> Hi,
> 
> Is there support for RFC3578 in FreeSWITCH? If not, is there any plans
> to implement it?
> 
> As far as I understand from quick browsing through this RFC, it is
> about mapping of ISUP overlap signalling to SIP protocol. In short,
> remote device (softswitch for example) sends short (incomplete) number
> in INVITE message to freeswitch, which responds with '484 address
> incomplete', then softswitch send another INVITE with more digits.
> Freeswitch then can send another 484, and after some time (enough
> digits collected), proceed with usual call establishment.
> 
> -- 
> Timur Irmatov, xmpp:irmatov at jabber.ru
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Fri Mar 12 07:55:29 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 12 Mar 2010 07:55:29 -0800
Subject: [Freeswitch-users] UK ring tone
Message-ID: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>

You can set it in the dialplan prior to transferring or bridging the  
incoming call.

Sent from my iPhone

On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> I've checked the internal and external sip profiles and think this
> needs to be set in autoload switch file as per:
>
> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
>
> I want by default a UK ringtone to be heard when I call a DDI that is
> configured on the system. At the moment it sounds like I'm calling the
> US.
>
> Thanks.
>
> -- http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From ivdreg at gmail.com  Fri Mar 12 08:44:57 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 12 Mar 2010 18:44:57 +0200
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
Message-ID: <b437a0e1003120844m52404de8l12c4d8e3299c54ac@mail.gmail.com>

Hi Brian and everyone in FS list,

Believe or not in some countries FAX is legal document in court and email is
not. As a tech guy I hate faxing but I live in world full with lawyer and
again because I am tech person when faxing is broken I have problems :(

P.S. This is just an opinion and not related to this lovely project ;)

2010/3/9 Brian West <brian at freeswitch.org>

> There is no such thing as security when humans are involved in the relay of
> information.  Thats fact.
>
> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
> involved its not secure.
>
> /b
>
> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>
> > While I'd love to see FAXing just die, especially with the amount of
> headache associated with it, the reality is that specific organizations
> still rely on it heavily on a daily basis (ie Health Care and
> Banking/Finance Industry). Plus they can legally get away with FAXing
> personal data directly between machines, since the FAX transmission doesn't
> seem to be directly covered by all their respective privacy regultions (ie
> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
> of these organizations just don't have the internal expertise to setup
> end-to-end encrypted email nor the impetus to deal with it, since FAXing
> works fine in their eyes. I think it's going to be around quite a while
> longer unless they actually stop making the FAX machines and force people to
> use something else.
> >
> > Cheers,
> > Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ivanov.maxim at gmail.com  Fri Mar 12 09:30:34 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 12 Mar 2010 17:30:34 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com> 
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com> 
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
Message-ID: <fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>

2010/2/26 Kristian Kielhofner <kristian.kielhofner at gmail.com>:
> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes

Doesn't help =( After getting first USER_BUSY it still makes call via
second part of dialstring.

originate {failure_causes=USER_BUSY}sofia/gateway/panas111/223|sofia/gateway/panas112/223
&playback(local_stream://moh)

Relevant log part is following:

2010-03-12 20:26:00.490396 [DEBUG] switch_ivr_originate.c:1176
variable string 0 = [failure_causes=USER_BUSY]
2010-03-12 20:26:00.494260 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/223 [35e8382d-e3fd-4190-b625-47f61ae6be16]
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:3020
(sofia/internal/223) State Change CS_NEW -> CS_INIT
freeswitch at internal> 2010-03-12 20:26:00.494260 [DEBUG]
switch_core_session.c:1000 Send signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_INIT
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:83 sofia/internal/223
SOFIA INIT
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:111
(sofia/internal/223) State Change CS_INIT -> CS_ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT going to sleep
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:130 sofia/internal/223
SOFIA ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING going to sleep
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA going to sleep
2010-03-12 20:26:00.494260 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.517975 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.550450 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [terminated][486]
2010-03-12 20:26:00.550450 [NOTICE] sofia.c:4149 Hangup
sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
2010-03-12 20:26:00.550450 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/223 [KILL]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_HANGUP
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:318 sofia/internal/223
Overriding SIP cause 486 with 486 from the other leg
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:350 Channel
sofia/internal/223 hanging up, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:46
sofia/internal/223 Standard HANGUP, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP going to sleep
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:53
sofia/internal/223 Standard REPORTING, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING going to sleep
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1137 Session
44644 (sofia/internal/223) Locked, Waiting on external entities
2010-03-12 20:26:00.550450 [DEBUG] switch_ivr_originate.c:2467
Originate Resulted in Error Cause: 17 [USER_BUSY]
2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1155 Session
44644 (sofia/internal/223) Ended
2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1157 Close
Channel sofia/internal/223 [CS_DESTROY]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/223) Running State Change CS_DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:267 sofia/internal/223
SOFIA DESTROY
2010-03-12 20:26:00.550450 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/223 [c8fdd3e0-d905-4f02-8955-d21d0bddf925]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:60
sofia/internal/223 Standard DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY going to sleep
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:3020
(sofia/internal/223) State Change CS_NEW -> CS_INIT
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_INIT
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:83 sofia/internal/223
SOFIA INIT
2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:111
(sofia/internal/223) State Change CS_INIT -> CS_ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.554707 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT going to sleep
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING
2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:130 sofia/internal/223
SOFIA ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING going to sleep
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA going to sleep
2010-03-12 20:26:00.578400 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.614451 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [terminated][486]
2010-03-12 20:26:00.614451 [NOTICE] sofia.c:4149 Hangup
sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
2010-03-12 20:26:00.614451 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/223 [KILL]
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_HANGUP
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP
2010-03-12 20:26:00.614451 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:318 sofia/internal/223
Overriding SIP cause 486 with 486 from the other leg
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:350 Channel
sofia/internal/223 hanging up, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:46
sofia/internal/223 Standard HANGUP, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP going to sleep
2010-03-12 20:26:00.618419 [DEBUG] switch_ivr_originate.c:2467
Originate Resulted in Error Cause: 17 [USER_BUSY]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:53
sofia/internal/223 Standard REPORTING, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING going to sleep
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1137 Session
44645 (sofia/internal/223) Locked, Waiting on external entities
2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1155 Session
44645 (sofia/internal/223) Ended
2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1157 Close
Channel sofia/internal/223 [CS_DESTROY]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/223) Running State Change CS_DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:267 sofia/internal/223
SOFIA DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:60
sofia/internal/223 Standard DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY going to sleep


From anthony.minessale at gmail.com  Fri Mar 12 09:40:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 11:40:09 -0600
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
	<fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>
Message-ID: <191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>

you need fail_on_single_reject=true inside the {} as well


On Fri, Mar 12, 2010 at 11:30 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> 2010/2/26 Kristian Kielhofner <kristian.kielhofner at gmail.com>:
> > http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>
> Doesn't help =( After getting first USER_BUSY it still makes call via
> second part of dialstring.
>
> originate
> {failure_causes=USER_BUSY}sofia/gateway/panas111/223|sofia/gateway/panas112/223
> &playback(local_stream://moh)
>
> Relevant log part is following:
>
> 2010-03-12 20:26:00.490396 [DEBUG] switch_ivr_originate.c:1176
> variable string 0 = [failure_causes=USER_BUSY]
> 2010-03-12 20:26:00.494260 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/223 [35e8382d-e3fd-4190-b625-47f61ae6be16]
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:3020
> (sofia/internal/223) State Change CS_NEW -> CS_INIT
> freeswitch at internal> 2010-03-12 20:26:00.494260 [DEBUG]
> switch_core_session.c:1000 Send signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_INIT
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:83 sofia/internal/223
> SOFIA INIT
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:111
> (sofia/internal/223) State Change CS_INIT -> CS_ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:130 sofia/internal/223
> SOFIA ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.517975 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.550450 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [terminated][486]
> 2010-03-12 20:26:00.550450 [NOTICE] sofia.c:4149 Hangup
> sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_channel.c:1896 Send signal
> sofia/internal/223 [KILL]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_HANGUP
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:318 sofia/internal/223
> Overriding SIP cause 486 with 486 from the other leg
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:350 Channel
> sofia/internal/223 hanging up, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/223 Standard HANGUP, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/223 Standard REPORTING, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:319
> (sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1137 Session
> 44644 (sofia/internal/223) Locked, Waiting on external entities
> 2010-03-12 20:26:00.550450 [DEBUG] switch_ivr_originate.c:2467
> Originate Resulted in Error Cause: 17 [USER_BUSY]
> 2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1155 Session
> 44644 (sofia/internal/223) Ended
> 2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1157 Close
> Channel sofia/internal/223 [CS_DESTROY]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/223) Running State Change CS_DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:267 sofia/internal/223
> SOFIA DESTROY
> 2010-03-12 20:26:00.550450 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/223 [c8fdd3e0-d905-4f02-8955-d21d0bddf925]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/223 Standard DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:3020
> (sofia/internal/223) State Change CS_NEW -> CS_INIT
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_INIT
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:83 sofia/internal/223
> SOFIA INIT
> 2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:111
> (sofia/internal/223) State Change CS_INIT -> CS_ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.554707 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT going to sleep
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:130 sofia/internal/223
> SOFIA ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING going to sleep
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA going to sleep
> 2010-03-12 20:26:00.578400 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.614451 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [terminated][486]
> 2010-03-12 20:26:00.614451 [NOTICE] sofia.c:4149 Hangup
> sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_channel.c:1896 Send signal
> sofia/internal/223 [KILL]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_HANGUP
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:318 sofia/internal/223
> Overriding SIP cause 486 with 486 from the other leg
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:350 Channel
> sofia/internal/223 hanging up, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/223 Standard HANGUP, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP going to sleep
> 2010-03-12 20:26:00.618419 [DEBUG] switch_ivr_originate.c:2467
> Originate Resulted in Error Cause: 17 [USER_BUSY]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/223 Standard REPORTING, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING going to sleep
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:319
> (sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1137 Session
> 44645 (sofia/internal/223) Locked, Waiting on external entities
> 2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1155 Session
> 44645 (sofia/internal/223) Ended
> 2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1157 Close
> Channel sofia/internal/223 [CS_DESTROY]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/223) Running State Change CS_DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:267 sofia/internal/223
> SOFIA DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/223 Standard DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY going to sleep
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ivanov.maxim at gmail.com  Fri Mar 12 10:46:06 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 12 Mar 2010 18:46:06 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com> 
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com> 
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com> 
	<fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com> 
	<191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>
Message-ID: <fbfa7b0c1003121046t3091f1d1l1dc6d1815a766eb8@mail.gmail.com>

> you need fail_on_single_reject=true inside the {} as well

brilliant! works like a charm!


From sean at obscuradigital.com  Fri Mar 12 14:18:00 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Fri, 12 Mar 2010 14:18:00 -0800
Subject: [Freeswitch-users] DTMF problem
Message-ID: <C7BFFD98.64D5%sean@obscuradigital.com>

    Question:

I have FS 1.0.5 installed on a Centos 5.4 server box.  I hope there?s a
simple answer to this question.

I?m having problems in particular when connecting to a outside conference
line on a completely different phone system entering numbers that the
receiver should accept.  For example, call into a conference line,
auto-attendent ask for the pin number, I then enter pin number, but The
receiving ivr tells me I?ve entered the wrong pin number.  I?m assuming that
I?ve reached a timeout and the ivr is letting me know that I?ve entered the
incorrect pin number.  My thinking is that this is a DTMF issue.  I?m new to
FS so not sure if I need to enable something in my dialplan or not.

Thoughts?
Thanks in advance
Sean
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From anthony.minessale at gmail.com  Fri Mar 12 14:32:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 16:32:48 -0600
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <C7BFFD98.64D5%sean@obscuradigital.com>
References: <C7BFFD98.64D5%sean@obscuradigital.com>
Message-ID: <191c3a031003121432l6fa95adcvc1693a268f9b289@mail.gmail.com>

Yet its quite possible.  If you are using a provider it can be a problem
with them as well.
There are numerous methods to debug DTMF issues on our wiki page.


On Fri, Mar 12, 2010 at 4:18 PM, Sean Holt <sean at obscuradigital.com> wrote:

>     Question:
>
> I have FS 1.0.5 installed on a Centos 5.4 server box.  I hope there?s a
> simple answer to this question.
>
> I?m having problems in particular when connecting to a outside conference
> line on a completely different phone system entering numbers that the
> receiver should accept.  For example, call into a conference line,
> auto-attendent ask for the pin number, I then enter pin number, but The
> receiving ivr tells me I?ve entered the wrong pin number.  I?m assuming that
> I?ve reached a timeout and the ivr is letting me know that I?ve entered the
> incorrect pin number.  My thinking is that this is a DTMF issue.  I?m new to
> FS so not sure if I need to enable something in my dialplan or not.
>
> Thoughts?
> Thanks in advance
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gavin.henry at gmail.com  Fri Mar 12 14:38:17 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Mar 2010 22:38:17 +0000
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
Message-ID: <13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>

Ok, not system wide?

Thanks,

Gavin.

On 12/03/2010, Michael S  Collins <msc at freeswitch.org> wrote:
> You can set it in the dialplan prior to transferring or bridging the
> incoming call.
>
> Sent from my iPhone
>
> On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> Hi,
>>
>> I've checked the internal and external sip profiles and think this
>> needs to be set in autoload switch file as per:
>>
>> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
>>
>> I want by default a UK ringtone to be heard when I call a DDI that is
>> configured on the system. At the moment it sounds like I'm calling the
>> US.
>>
>> Thanks.
>>
>> -- http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From anthony.minessale at gmail.com  Fri Mar 12 15:05:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 17:05:40 -0600
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
Message-ID: <191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>

you can set it globally in vars.xml too


On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Ok, not system wide?
>
> Thanks,
>
> Gavin.
>
> On 12/03/2010, Michael S  Collins <msc at freeswitch.org> wrote:
> > You can set it in the dialplan prior to transferring or bridging the
> > incoming call.
> >
> > Sent from my iPhone
> >
> > On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:
> >
> >> Hi,
> >>
> >> I've checked the internal and external sip profiles and think this
> >> needs to be set in autoload switch file as per:
> >>
> >> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
> >>
> >> I want by default a UK ringtone to be heard when I call a DDI that is
> >> configured on the system. At the moment it sounds like I'm calling the
> >> US.
> >>
> >> Thanks.
> >>
> >> -- http://www.suretecsystems.com/services/openldap/
> >> http://www.suretectelecom.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> Sent from my mobile device
>
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Fri Mar 12 15:09:29 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Mar 2010 17:09:29 -0600
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
	<191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
Message-ID: <FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>

in fact an example for uk-ring is already there.

/b

On Mar 12, 2010, at 5:05 PM, Anthony Minessale wrote:

> you can set it globally in vars.xml too
> 
> 
> On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
> Ok, not system wide?
> 
> Thanks,
> 
> Gavin.

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From m.sobkow at marketelsystems.com  Fri Mar 12 15:48:25 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 12 Mar 2010 17:48:25 -0600
Subject: [Freeswitch-users] OpenZap/Sangoma drivers
Message-ID: <4B9AD2C9.1010004@marketelsystems.com>

I filed a bug report this morning because of a Freeswitch build problem, 
but the resolution was to update the OpenZap/Sangoma drivers.  Updating 
the Sangoma drivers was a little hairy because of our project build 
process needing to be updated, but I got it figured out and got the 
latest build of Freeswitch installed (svn checkout this morning.)  Next 
I need to re-test the call recordings.


From fs-list at communicatefreely.net  Fri Mar 12 18:38:08 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 12 Mar 2010 21:38:08 -0500
Subject: [Freeswitch-users] voicemail_greeting_number stickiness
Message-ID: <4B9AFA90.6050409@communicatefreely.net>


Hello all,

I'm trying to set up our voice mail system so that greetings can be mapped to failure conditions,
ie. "I'm in my office, but not taking calls" when a phone is busy or in DND mode, but "Sorry I'm not
here" if no_answer.

I programmed things to set voicemail_greeting_number in the dialplan based on the bridge failure
cause set when we rang the phone, but when I execute the voicemail application, it stores that
greeting number in the database.  I would much rather override for this call only, so that the user
can still take advantage of the voicemail application's option to select a greeting.  (which would
be used as the no answer greeting under normal circumstances).

Any thoughts?

-Tim


From jmesquita at freeswitch.org  Fri Mar 12 20:23:19 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Mar 2010 01:23:19 -0300
Subject: [Freeswitch-users] voicemail_greeting_number stickiness
In-Reply-To: <4B9AFA90.6050409@communicatefreely.net>
References: <4B9AFA90.6050409@communicatefreely.net>
Message-ID: <ebbc2d551003122023r22c0ed0bx580dad9893e21a1c@mail.gmail.com>

Use hangup hook to set it back to whatever is default?

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Fri, Mar 12, 2010 at 11:38 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

>
> Hello all,
>
> I'm trying to set up our voice mail system so that greetings can be mapped
> to failure conditions,
> ie. "I'm in my office, but not taking calls" when a phone is busy or in DND
> mode, but "Sorry I'm not
> here" if no_answer.
>
> I programmed things to set voicemail_greeting_number in the dialplan based
> on the bridge failure
> cause set when we rang the phone, but when I execute the voicemail
> application, it stores that
> greeting number in the database.  I would much rather override for this
> call only, so that the user
> can still take advantage of the voicemail application's option to select a
> greeting.  (which would
> be used as the no answer greeting under normal circumstances).
>
> Any thoughts?
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fs-list at communicatefreely.net  Fri Mar 12 21:34:50 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sat, 13 Mar 2010 00:34:50 -0500
Subject: [Freeswitch-users] Gateway context for incoming calls
Message-ID: <4B9B23FA.6080103@communicatefreely.net>

Hello,

Is there a way for incoming calls from a gateway to go to a context other than the default one set
in the sofia profile?  I need a way to separate calls from providers vs. calls from the public at large.

Here's the scenario:

Our external sofia profile maps to the public IP address and listens on port 5060.  Our providers
send calls to this address, but we also need to have anonymous sip calls come in to the public
address, on the default port.  We want the calls from our SIP trunks to get billed appropriately,
and increment our mod_limit counters, etc.  If a call comes in from somewhere other than a SIP
trunk, we want it to have access to extensions, but be rated as a free call.

All our SIP trunks are mapped using static addresses - there isn't any registration or
authentication happening in either direction.  Each one has a gateway entry.

Thanks!

-Tim


From msc at freeswitch.org  Fri Mar 12 22:59:20 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Mar 2010 22:59:20 -0800
Subject: [Freeswitch-users] Gateway context for incoming calls
In-Reply-To: <4B9B23FA.6080103@communicatefreely.net>
References: <4B9B23FA.6080103@communicatefreely.net>
Message-ID: <87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>

Try sending one of these calls to the info application in your public
dialplan. Most likely there will be a few fields which you can key off of in
your dialplan conditions. If your providers have static IPs then you can
always use the network_addr field to match on. Once you match on a
particular IP address (or whichever field) then you can execute the transfer
app and send the call to another context altogether:

<extension name="example IP addr match">
    <condition field="network_addr" expression="^1\.2\.3\.4$"/> <!-- IP addr
of originating server -->
    <condition field="destination_number" expression="^(.*)$">
         <action application="transfer" data="$1 XML my_context"/>
    </condition>
</extension>

Let us know how it goes.
-MC

On Fri, Mar 12, 2010 at 9:34 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> Is there a way for incoming calls from a gateway to go to a context other
> than the default one set
> in the sofia profile?  I need a way to separate calls from providers vs.
> calls from the public at large.
>
> Here's the scenario:
>
> Our external sofia profile maps to the public IP address and listens on
> port 5060.  Our providers
> send calls to this address, but we also need to have anonymous sip calls
> come in to the public
> address, on the default port.  We want the calls from our SIP trunks to get
> billed appropriately,
> and increment our mod_limit counters, etc.  If a call comes in from
> somewhere other than a SIP
> trunk, we want it to have access to extensions, but be rated as a free
> call.
>
> All our SIP trunks are mapped using static addresses - there isn't any
> registration or
> authentication happening in either direction.  Each one has a gateway
> entry.
>
> Thanks!
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Sat Mar 13 00:51:13 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Mar 2010 19:51:13 +1100
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
Message-ID: <20100313085113.GA32174@jdc.jasonjgw.net>

Code has now been posted at https://developer.skype.com/silk/

It appears from that page that there may be patent issues for anything other
than testing of the codec, although it is stated elsewhere that the licence
terms are royalty-free.

To build the module (for those planning to do the aforementioned testing) you
need to run make in libs/silk first to compile the library, then build
FreeSWITCH; this will change once the build system is updated appropriately.



From steveu at coppice.org  Sat Mar 13 02:16:51 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 13 Mar 2010 18:16:51 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <20100313085113.GA32174@jdc.jasonjgw.net>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>	<20100310051344.GA19653@jdc.jasonjgw.net>	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
	<20100313085113.GA32174@jdc.jasonjgw.net>
Message-ID: <4B9B6613.4040901@coppice.org>

On 03/13/2010 04:51 PM, Jason White wrote:
> Code has now been posted at https://developer.skype.com/silk/
>
> It appears from that page that there may be patent issues for anything other
> than testing of the codec, although it is stated elsewhere that the licence
> terms are royalty-free.
>    
That sounds like they are trying for a similar arrangement Polycom has 
with G.722.1. You can use G.722.1 royalty free, if you ask for a licence 
first and comply with some simple conditions - mostly ensuring that you 
properly follow the G.722.1 spec. It works well enough for most 
purposes, though it conflicts with some free licences, and there have 
been some disagreements. For example, Polycom said it was OK for 
Freeswitch to be distributed with the source code for G.722.1 bundled. 
It seems Digium believe Asterisk may not be distributed with the source 
code for G.722.1.
> To build the module (for those planning to do the aforementioned testing) you
> need to run make in libs/silk first to compile the library, then build
> FreeSWITCH; this will change once the build system is updated appropriately.
>
>    
Steve



From brian at freeswitch.org  Sat Mar 13 08:08:37 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 10:08:37 -0600
Subject: [Freeswitch-users] Gateway context for incoming calls
In-Reply-To: <87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>
References: <4B9B23FA.6080103@communicatefreely.net>
	<87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>
Message-ID: <47ED6B59-7300-467B-B879-643266018395@freeswitch.org>

Its worse.. the only way you can map an inbound call back to the gateway in question is the far side has to call the registered contact you sent on outbound registration.  Most services don't follow this rule.

/b

On Mar 13, 2010, at 12:59 AM, Michael Collins wrote:

> Try sending one of these calls to the info application in your public dialplan. Most likely there will be a few fields which you can key off of in your dialplan conditions. If your providers have static IPs then you can always use the network_addr field to match on. Once you match on a particular IP address (or whichever field) then you can execute the transfer app and send the call to another context altogether:
> 
> <extension name="example IP addr match">
>     <condition field="network_addr" expression="^1\.2\.3\.4$"/> <!-- IP addr of originating server -->
>     <condition field="destination_number" expression="^(.*)$">
>          <action application="transfer" data="$1 XML my_context"/>
>     </condition>
> </extension>
> 
> Let us know how it goes.
> -MC



From gavin.henry at gmail.com  Sat Mar 13 11:20:31 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Mar 2010 19:20:31 +0000
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
	<191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
	<FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>
Message-ID: <13ca621c1003131120n4a322ad1mfa7b910f593b7b5c@mail.gmail.com>

Ok, thanks. I think there is a us and fr one by default. Will comment
out those and test.

Thanks,

Gavin.

On 12/03/2010, Brian West <brian at freeswitch.org> wrote:
> in fact an example for uk-ring is already there.
>
> /b
>
> On Mar 12, 2010, at 5:05 PM, Anthony Minessale wrote:
>
>> you can set it globally in vars.xml too
>>
>>
>> On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com>
>> wrote:
>> Ok, not system wide?
>>
>> Thanks,
>>
>> Gavin.
>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From jbrucehopkins at gmail.com  Sat Mar 13 14:56:50 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sat, 13 Mar 2010 22:56:50 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <000001cabfbb$f2114c90$d633e5b0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
Message-ID: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>

Hi Roly,

I promised I'd let you know when I got it working - so here it is:

As Brian said, doublenat is no longer needed.

It looks like I must have just been confusing things horribly by using
various things I'd read about, including:

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

<param name="aggressive-nat-detection" value="true"/>


These, and who knows what other mucking about I had been doing, must have
confused everything as none of this is now necessary.


All I did - after a reinstall - to get everything working through double
NATs is:


1.  In vars.xml

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>


 <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>


2.  Then in prefix/sip_profiles/internal.xml

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

 <param name="ext-sip-ip" value="$${external_sip_ip)"/>


Then as the man says, it just works.

I decided not to enable options pings to keep the far-end NAT open to enable
calling the remote phone - instead I configured it to send keep-alives.


And that's it.  I can't believe the knots I was tying myself in.  It just
works !

cheers

Bruce








On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Mar 13 17:05:40 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 19:05:40 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
Message-ID: <3B76DFA1-6304-4B84-A478-387C1CED9D3C@freeswitch.org>

Its a common thing to try too hard.  I personally put in countless hours to make this stuff JUST work. ;)  Also if you're traversing the nat and you expect to talk to clients behind the nat and outside the nat on the same profile you will be required to prefix both ext-rtp-ip and ext-sip-ip with "autonat:<my.fs.ext.ip>" and make sure local-network-acl is set to localnet.auto... I think thats the one in the defaults too.  This gives FreeSWITCH the ability to dynamically lie to the client about which IP is which so that the same profile can talk to clients on either side of the nat.

/b

On Mar 13, 2010, at 4:56 PM, Bruce Hopkins wrote:

> And that's it.  I can't believe the knots I was tying myself in.  It just works !



From mattdfong at gmail.com  Sat Mar 13 17:24:48 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 13 Mar 2010 17:24:48 -0800
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback 	doesn't seem to get called
In-Reply-To: <5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
	<5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
Message-ID: <4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>

Hi Michael,

the cell phone carriers in the US use a beep tone that is out of the range
that is typically detected. i've had success using tone_detect rather than
mod_vmd. if my memory serves me correctly 786 is the tone you need to listen
for. the same is true for sprint and t-mobile voice mail beeps

--matt
freeswitch based voice broadcasting -
http://www.hellohunter.com/voice_broadcast.php
freeswitch based predictive dialing -
http://www.hellohunter.com/predictive_dialer.php

On Wed, Mar 3, 2010 at 7:04 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Has anyone experienced issues with vmd or dtmf with Verizon wireless?  I
> have another script that doesn't even register digits pressed -- it's like
> no keys are pressed at all.
>
> I've checked the FS log and it doesn't seem to log anything with regard to
> the mod_vmd other than an indication of MIN_TIME of 8000 (not sure what that
> does exactly).
>
> Does anyone have any sample scripts available other than what's on the FS
> wiki?
>
>
> On Mon, Mar 1, 2010 at 10:17 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> I've got the following Lua script working in a sense, but the
>> InputCallback never seems to get called while the file is being streamed to
>> the call recipient.  I've tried moving the "vmd start" command and set input
>> callback around a bit, but to no avail.  I'm testing this against a cell
>> phone voice mailbox (Verizon).
>>
>>
>>> freeswitch.consoleLog("info","########################################################\n\n");
>>>
>>>  number_to_call = argv[1]
>>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>>
>>> voicemail_detected = false;
>>>
>>> function onInput(s, type, obj)
>>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>>> *************\n");
>>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>>> session:getVariable("vmd_detect") .. " *************\n");
>>>
>>>      if(type == "event" and voicemail_detected == false) then
>>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>>> MACHINE DETECTED *************\n");
>>>         voicemail_detected = true;
>>>         return "break";
>>>     end
>>> end
>>>
>>> function playbackMessage()
>>>     sleep_time = 1000;
>>>     if(voicemail_detected) then
>>>        sleep_time = 2500;
>>>     end
>>>     -- sleep a second
>>>     session:sleep(sleep_time);
>>>     -- play a file
>>>     session:streamFile(message_to_play);
>>> end
>>>
>>> function notify()
>>>     session =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>>> .. number_to_call)
>>>
>>>
>>>
>>>     if(session:ready()) then
>>>         -- answer the call
>>>         session:answer();
>>>         session:setInputCallback("onInput", "");
>>>         session:execute("vmd","start");
>>>
>>>         playbackMessage();
>>>         if(voicemail_detected) then
>>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>>             playbackMessage();
>>>         end
>>>
>>>         freeswitch.consoleLog("notice", "********* hanging up session
>>> **********\n");
>>>         -- hangup
>>>         session:hangup();
>>>     end
>>> end
>>>
>>> notify();
>>>
>>>
>>> freeswitch.consoleLog("info","########################################################\n\n");
>>>
>>
>> --
>> Michael De Lorenzo
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Sat Mar 13 17:53:20 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 12:53:20 +1100
Subject: [Freeswitch-users] not answering calls on TDM400 port
Message-ID: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>

Hello,

I noticed Freeswitch wasn't answering calls on one of the TDM400 ports
yesterday.

All other ports seemed to be OK.

Nothing showed up on the console, set at debug log level.

I restarted Freeswitch and it came good.

What should I do if it happens again? Is there any debugging
information I can extract from the system? If so how?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Sat Mar 13 18:47:21 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 20:47:21 -0600
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback  doesn't seem to get called
In-Reply-To: <4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
	<5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
	<4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>
Message-ID: <4EC6125F-683B-409D-B7D6-C8842586BAD2@freeswitch.org>

You'll need to file a jira.  Include a script that exhibits the behavior.  You'll also need to document it fully and the dialplan entry you're using to produce this issue.

/b

On Mar 13, 2010, at 7:24 PM, Matthew Fong wrote:

> Hi Michael,
> 
> the cell phone carriers in the US use a beep tone that is out of the range that is typically detected. i've had success using tone_detect rather than mod_vmd. if my memory serves me correctly 786 is the tone you need to listen for. the same is true for sprint and t-mobile voice mail beeps
> 
> --matt
> freeswitch based voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
> freeswitch based predictive dialing - http://www.hellohunter.com/predictive_dialer.php

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From brian at microcomaustralia.com.au  Sat Mar 13 20:02:25 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:02:25 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
	<3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
	<cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>
Message-ID: <3c5cf5261003132002v715c25a3o945baf6062f4f20@mail.gmail.com>

On 1 March 2010 19:10, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
> On this one, I would have to check in the code. I don't know if the span
> order here makes a difference (my guess is no).

Difference isn't so much the order, but one block has tonegroup set (I
assume this is for an extension), and the other doesn't.
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Mar 13 20:11:07 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:11:07 +1100
Subject: [Freeswitch-users] internal/external profiles
In-Reply-To: <87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>
References: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
	<87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>
Message-ID: <3c5cf5261003132011l627f91a8wd953aeb428493fca@mail.gmail.com>

On 25 February 2010 03:18, Michael Collins <msc at freeswitch.org> wrote:
> No problem. Separating them has a few advantages: security, scalability, and
> readability. The first one on the list is definitely the most important. If
> you stuff everything in the internal profile it's easier to open yourself up
> to toll fraud.

Hello,

I am still confused as to how using the external profile can help
reduce toll fraud.

On my system both internal and external have:

<param name="context" value="public"/>

Also, what happens when freeswitch receives an anonymous connection?
Are anonymous connections allowed or rejected? Where is that behaviour
controlled?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Sat Mar 13 20:15:00 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:15:00 +1100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <4B8A300C.4060805@xpirio.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
	<4B8A300C.4060805@xpirio.com>
Message-ID: <3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>

2010/2/28 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> problem solved with -vm
> with this option we now have the usual low load for 50-70 conference users

I can't see any documentation for vm, either here:

http://wiki.freeswitch.org/wiki/Command_line

or here:

huey:/opt/freeswitch/conf# /opt/freeswitch/bin/freeswitch --help
these are the optional arguments you can pass to freeswitch
	-nf                    -- no forking
	-u [user]              -- specify user to switch to
	-g [group]             -- specify group to switch to
	-help                  -- this message
	-version               -- print the version and exit
	-waste                 -- allow memory waste
	-core                  -- dump cores
	-hp                    -- enable high priority settings
	-vg                    -- run under valgrind
	-nosql                 -- disable internal sql scoreboard
	-heavy-timer           -- Heavy Timer, possibly more accurate but at a cost
	-nonat                 -- disable auto nat detection
	-nocal                 -- disable clock calibration
	-nort                  -- disable clock clock_realtime
	-stop                  -- stop freeswitch
	-nc                    -- do not output to a console and background
	-c                     -- output to a console and stay in the foreground
	-conf [confdir]        -- specify an alternate config dir
	-log [logdir]          -- specify an alternate log dir
	-run [rundir]          -- specify an alternate run dir
	-db [dbdir]            -- specify an alternate db dir
	-mod [moddir]          -- specify an alternate mod dir
	-htdocs [htdocsdir]    -- specify an alternate htdocs dir
	-scripts [scriptsdir]  -- specify an alternate scripts dir



What does it do?
-- 
Brian May <brian at microcomaustralia.com.au>


From sos at sokhapkin.dyndns.org  Sat Mar 13 20:26:28 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 13 Mar 2010 23:26:28 -0500
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>
References: <4B857226.10308@xpirio.com> <4B8A300C.4060805@xpirio.com>
	<3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>
Message-ID: <201003132326.28390.sos@sokhapkin.dyndns.org>

-vm option has been removed and is the default behavior now, -heavy-timer 
option switches to old (pre-1.0.5) way.

On Saturday 13 March 2010, Brian May wrote:
> 2010/2/28 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> > problem solved with -vm
> > with this option we now have the usual low load for 50-70 conference
> > users
> 
> I can't see any documentation for vm, either here:
> 
> http://wiki.freeswitch.org/wiki/Command_line
> 
> or here:
> 
> huey:/opt/freeswitch/conf# /opt/freeswitch/bin/freeswitch --help
> these are the optional arguments you can pass to freeswitch
> 	-nf                    -- no forking
> 	-u [user]              -- specify user to switch to
> 	-g [group]             -- specify group to switch to
> 	-help                  -- this message
> 	-version               -- print the version and exit
> 	-waste                 -- allow memory waste
> 	-core                  -- dump cores
> 	-hp                    -- enable high priority settings
> 	-vg                    -- run under valgrind
> 	-nosql                 -- disable internal sql scoreboard
> 	-heavy-timer           -- Heavy Timer, possibly more accurate but at a
>  cost -nonat                 -- disable auto nat detection
> 	-nocal                 -- disable clock calibration
> 	-nort                  -- disable clock clock_realtime
> 	-stop                  -- stop freeswitch
> 	-nc                    -- do not output to a console and background
> 	-c                     -- output to a console and stay in the foreground
> 	-conf [confdir]        -- specify an alternate config dir
> 	-log [logdir]          -- specify an alternate log dir
> 	-run [rundir]          -- specify an alternate run dir
> 	-db [dbdir]            -- specify an alternate db dir
> 	-mod [moddir]          -- specify an alternate mod dir
> 	-htdocs [htdocsdir]    -- specify an alternate htdocs dir
> 	-scripts [scriptsdir]  -- specify an alternate scripts dir
> 
> 
> 
> What does it do?
> 



From nepaligas at yahoo.com  Sat Mar 13 22:22:14 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Sat, 13 Mar 2010 22:22:14 -0800 (PST)
Subject: [Freeswitch-users] problem in sending calls to internal Gateway.
Message-ID: <249845.71206.qm@web65402.mail.ac4.yahoo.com>

hello,

My extension are working properly. But they can't send calls to gateway. I have 2 interfaces, one with public and another with private lan. Gateway is in private lan. Please suggest.
My configurations are as follows:

/conf/sip_profiles/external.xml:
    <param name="rtp-ip" value="110.4.12.111"/>
    <param name="sip-ip" value="110.4.12.111"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

/conf/sip_profiles/internal.xml:
     <param name="rtp-ip" value="192.168.1.11"/>
     <param name="sip-ip" value="192.168.1.11"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="nat.auto"/>

haven't made any changes in conf/autoload_configs/acl.conf.xml

to route calls from providers, /dialplan/public/providers.xml:
    <extension name="provider">
    <condition field="network_addr" expression="200.144.44.177"/>
    <condition field="destination_number" expression="^(\d+)$">
        <action application="transfer" data="$1 XML context_AX"/>
    </condition>
   </extension>

gateway is defined at /conf/sip_profiles/internal/gateways.xml:
   <include>
   <gateway name="gateway_AX">
    <param name="username" value="admin"/>
    <param name="password" value="nepal123"/>
    <param name="proxy" value="192.168.1.100"/>
    <param name="register" value="false"/>
    <param name="extension" value="1001"/>
    </gateway>
   </include>


To route internal calls from extensions, /conf/dialplan/default.xml:
<extension name="my_internal">
    <!--<condition field="${ruleset}" expression="internal" /> -->
    <condition field="destination_number" expression="^(d+)$">
      <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
    </condition>
  </extension>

to route calls from providers, conf/dialplan/inbound_routing.xml:
<context name="context_AX">
    <extension name="from_provider_to_AX">    
    <!--<condition field="destination_number" expression="^00(1.*)$">-->
    <condition field="destination_number" expression="^9(\d+)$">    
    <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
    </condition>
    </extension>
</context>

Please help.


      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/



From dome at tel.co.th  Sun Mar 14 05:37:04 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 14 Mar 2010 19:37:04 +0700
Subject: [Freeswitch-users] Inline and cond
Message-ID: <8ccbff061003140537r550234d2j6c6124a6916ea8ec@mail.gmail.com>

I try to use cond in my dialplan

     <condition field="destination_number" expression="^(1111)">
        <action application="answer"/>
        <action inline="true" application="set" data="b=25"/>
     </condition>
     <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$">
        <action application="log" data="ALERT ${b} is indeed greater than 12"/>
        <anti-action application="log" data="ALERT ${b} is not greater
than 12"/>
     </condition>

it's work but when use set b=${b1} it's not work

     <condition field="destination_number" expression="^(1111)">
        <action application="answer"/>
        <action inline="true" application="set" data="b1=25"/>
        <action inline="true" application="set" data="b=${b1}"/>
     </condition>
     <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$">
        <action application="log" data="ALERT ${b} is indeed greater than 12"/>
        <anti-action application="log" data="ALERT ${b} is not greater
than 12"/>
     </condition>

What's wrong ? inline or cond ?

Dome C.


From yehavi.bourvine at gmail.com  Sun Mar 14 08:18:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 14 Mar 2010 17:18:23 +0200
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a call
	is left on hold?
Message-ID: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>

Hello,

   We have a strange behavior which my users told me today, but I have a
feeling it lasts for a long time (we are running now version 16972).

User A calls user B on a Polycom. User B press the "transfer" key and calls
C who answers. B then press "transfer" again, and now A talks to C. However,
B is left with a held call to the voicemail of C...

Anyone saw this behaviour?  unattended transfer and conference work ok.

                             Thanks! __Yehavi:
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From brian at freeswitch.org  Sun Mar 14 10:50:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:50:28 -0500
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
Message-ID: <C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>

Are you doing this with the latest SVN and default dialplan or are you doing something out of the usual?

/b

On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:

> Hello,
>  
>    We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972).
>  
> User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C...
>  
> Anyone saw this behaviour?  unattended transfer and conference work ok.
>  
>                              Thanks! __Yehavi: 



From brian at freeswitch.org  Sun Mar 14 10:53:55 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:53:55 -0500
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
Message-ID: <86173DCE-6BE0-4663-B90E-81244587F4C4@freeswitch.org>

I can confirm I get the proper behavior.

/b

On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:

> Hello,
>  
>    We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972).
>  
> User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C...
>  
> Anyone saw this behaviour?  unattended transfer and conference work ok.
>  
>                              Thanks! __Yehavi: 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From quentezcorp at gmail.com  Fri Mar 12 13:03:28 2010
From: quentezcorp at gmail.com (Quentin CALVEZ)
Date: Fri, 12 Mar 2010 22:03:28 +0100
Subject: [Freeswitch-users] js.dll crash on Windows Server 2008 R2 x64
Message-ID: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>

Hello everybody,

I've been trying Freeswitch those last days and there is one error that I
don't know what to do about. I'm using FusionPBX to manage my freeswitch
installation and it works great, however, whenever I try to use a function
involving javascript (auto atendants, hunt groups...) the server crashs
telling me that the faulty module is js32.dll.

I really don't know what to do about it. Could someone help me :) ?

Regards.

Quentin Calvez
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From roland at haenel.me  Sat Mar 13 12:38:26 2010
From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=)
Date: Sat, 13 Mar 2010 21:38:26 +0100
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
Message-ID: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>

 Hello,

I'm developing a very simple call-through application using FreeSWITCH, but
currently
I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
header
for outgoing calls if I want so signal an arbitrary caller ID in the 'From'
field.

Setup is as follows:
   - a-leg comes in from an external gateway
   - dialplan dispatches the call to a perl handler script
   - perl script looks like this (simplified as much as possible for this
description):

            $num = "02222222";
            $session->answer();
            $session->setVariable("effective_caller_id_name", "The
Redirector");
            $session->setVariable("sip_h_P-Asserted-Identity", '<
sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
            $session->execute("bridge", "sofia/gateway/mygateway/$num);

The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity
header as
set in the perl script. But there is already an existing P-Asserted-Identity
header by default,
so I end up with two of them. This is the outgoing INVITE (b-leg):


INVITE sip:02222222 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
Max-Forwards: 63
From: "The Redirector" <sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
>;tag=DBSvZ9NZc750g
To: <sip:02222222 at x.x.x.x>
Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
CSeq: 128071597 INVITE
Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 244
X-Port: 5060
X-FS-Support: update_display
P-Asserted-Identity: "The Redirector"
<sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>
P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
>

[... sdp ...]

Numbers are:

    01111111       - the original caller (a-leg & b-leg From:)
    02222222       - the number redirected to (b-leg To:)
    0123456789    - the number my provider needs in P-Asserted-Identity
('redirector's number')

So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
line, but unfortunately it already includes another P-Asserted-Identity
011111111
line, and that breaks the setup at my provider.

I really appreciate any help.

Greetings,
Roland

--
Roland Haenel
QSC AG - http://www.qsc.de
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From rentmycoder at gmail.com  Sun Mar 14 10:40:02 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Sun, 14 Mar 2010 18:40:02 +0100
Subject: [Freeswitch-users] how to access session variables after hangup???
Message-ID: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>

Hi guys,
Freeswitch rocks!
except the documentation:(
I'm not able to figure out how to access session variables after hangup...
I try to dial out on a gateway and detect it the called party hangs up
before bridge...

bgapi originate
{ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
hangup.lua ${uuid}'}sofia/gateway/phoneno &park()

I cannot access the variables using:
1. script arguments:
'luarun hangup.lua ${uuid}'}
2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
(${api_hangup_hook} contains a variable)

2. session:getVariable("originate_disposition")
2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
to call global 'getVar' (a nil value)

3. params:getHeader("variable_sip_req_uri")
2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
to index global 'params' (a nil value)

I have asked this before on the dev list too, but Anthony didn't gave
me a clean answer here:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html

"In the script you can access the data from the env event which is
automatically created for you and it
contains all the channel variables from that departed channel."

Ok, but how?????

Please help...


From spencer at 5ninesolutions.com  Sun Mar 14 10:49:42 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 10:49:42 -0700
Subject: [Freeswitch-users] 100% CPU
Message-ID: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>

Hi,
I have Freeswitch running as a SBC in front of several Asterisk  
machines and without warning one of the Freeswitch processes will use  
100% of one of the CPU cores and stop responding to SIP requests.  I'm  
using the latest svn on Centos 5.4 in a xen vm.  Basically everything  
will be fine for about 3-4 hours and then this happens.


[root at sip ~]# cat /proc/cpuinfo
processor	: 0
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 0
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 1
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 1
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 2
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 2
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 3
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 3
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

[root at sip ~]# uname -r
2.6.18-164.11.1.el5xen


--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980


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From brian at freeswitch.org  Sun Mar 14 10:58:35 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:58:35 -0500
Subject: [Freeswitch-users] js.dll crash on Windows Server 2008 R2 x64
In-Reply-To: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>
References: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>
Message-ID: <E0BD3C11-9978-4880-8885-0FEB2BE5DEF7@freeswitch.org>

We can't help you because you have failed to provide any information in context.

Please read this link http://wiki.freeswitch.org/wiki/Reporting_Bugs

Then collect the info and report it if the problem persists on latest SVN.

Thanks,
Brain

On Mar 12, 2010, at 3:03 PM, Quentin CALVEZ wrote:

> Hello everybody,
> 
> I've been trying Freeswitch those last days and there is one error that I don't know what to do about. I'm using FusionPBX to manage my freeswitch installation and it works great, however, whenever I try to use a function involving javascript (auto atendants, hunt groups...) the server crashs telling me that the faulty module is js32.dll.
> 
> I really don't know what to do about it. Could someone help me :) ?
> 
> Regards.
> 
> Quentin Calvez
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Mar 14 10:59:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:59:36 -0500
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <97634AC6-B936-40F3-9689-CC41E640D0FF@freeswitch.org>

Set the variable sip_cid_type=none and it will only attach the one you set.

/b

On Mar 13, 2010, at 2:38 PM, Roland H?nel wrote:

> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net>
> P-Asserted-Identity: <sip:0123456789 at q-loud.net>

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From brian at freeswitch.org  Sun Mar 14 11:00:10 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:00:10 -0500
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>
References: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>
Message-ID: <39B195A1-5A29-418C-A98C-6DCDC829FA39@freeswitch.org>

Pass them as arguments to your luarun hangup hook.

/b

On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:

> Hi guys,
> Freeswitch rocks!
> except the documentation:(
> I'm not able to figure out how to access session variables after hangup...
> I try to dial out on a gateway and detect it the called party hangs up
> before bridge...
> 
> bgapi originate
> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
> 
> I cannot access the variables using:
> 1. script arguments:
> 'luarun hangup.lua ${uuid}'}
> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> 
> 2. session:getVariable("originate_disposition")
> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
> to call global 'getVar' (a nil value)
> 
> 3. params:getHeader("variable_sip_req_uri")
> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
> to index global 'params' (a nil value)
> 
> I have asked this before on the dev list too, but Anthony didn't gave
> me a clean answer here:
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
> 
> "In the script you can access the data from the env event which is
> automatically created for you and it
> contains all the channel variables from that departed channel."
> 
> Ok, but how?????
> 
> Please help...
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Mar 14 11:00:55 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:00:55 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
Message-ID: <A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>

Are you on the latest SVN?

If the problem persists after you update to SVN trunk please read http://wiki.freeswitch.org/wiki/Reporting_Bugs and report the issue to jira.

/b

On Mar 14, 2010, at 12:49 PM, Spencer Thomason wrote:

> I have Freeswitch running as a SBC in front of several Asterisk machines and without warning one of the Freeswitch processes will use 100% of one of the CPU cores and stop responding to SIP requests.  I'm using the latest svn on Centos 5.4 in a xen vm.  Basically everything will be fine for about 3-4 hours and then this happens.
> 



From mrene_lists at avgs.ca  Sun Mar 14 11:02:13 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 14 Mar 2010 14:02:13 -0400
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <C6955D38-7F60-472F-A72F-52C9FAA43595@avgs.ca>

Hi,

If you want to manually add the P-Asserted-Identity header, you must  
first set sip_cid_type to none. Or you could just set  
effective_caller_id_number to the number that should show up in PAI.

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




On 13-Mar-10, at 3:38 PM, Roland H?nel wrote:

> Hello,
>
> I'm developing a very simple call-through application using  
> FreeSWITCH, but currently
> I'm stuck because my provider requires me to send a 'P-Asserted- 
> Identity' header
> for outgoing calls if I want so signal an arbitrary caller ID in the  
> 'From' field.
>
> Setup is as follows:
>    - a-leg comes in from an external gateway
>    - dialplan dispatches the call to a perl handler script
>    - perl script looks like this (simplified as much as possible for  
> this description):
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The  
> Redirector");
>             $session->setVariable("sip_h_P-Asserted-Identity", '<sip:0123456789 at q-loud.net 
> >');
>             $session->execute("bridge", "sofia/gateway/mygateway/ 
> $num);
>
> The problem is now that FreeSWITCH correctly insertes a P-Asserted- 
> Identity header as
> set in the perl script. But there is already an existing P-Asserted- 
> Identity header by default,
> so I end up with two of them. This is the outgoing INVITE (b-leg):
>
> INVITE sip:02222222 at x.x.x.x SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
> Max-Forwards: 63
> From: "The Redirector" <sip:01111111 at q-loud.net>;tag=DBSvZ9NZc750g
> To: <sip:02222222 at x.x.x.x>
> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
> CSeq: 128071597 INVITE
> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 244
> X-Port: 5060
> X-FS-Support: update_display
> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net>
> P-Asserted-Identity: <sip:0123456789 at q-loud.net>
>
> [... sdp ...]
> Numbers are:
>
>     01111111       - the original caller (a-leg & b-leg From:)
>     02222222       - the number redirected to (b-leg To:)
>     0123456789    - the number my provider needs in P-Asserted- 
> Identity ('redirector's number')
>
> So, the INVITE is correct, it includes my P-Asserted-Identity  
> 0123456789
> line, but unfortunately it already includes another P-Asserted- 
> Identity 011111111
> line, and that breaks the setup at my provider.
>
> I really appreciate any help.
>
> Greetings,
> Roland
>
> --
> Roland Haenel
> QSC AG - http://www.qsc.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lawwton at gmail.com  Sun Mar 14 11:06:59 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:06:59 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
Message-ID: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>

All:

We are currently exploring the possibility of using mod_nibblebill and
integrating it with a conference. It seems like it offers all the
features and more that we need. The way we create a conference now is
as follows:

originate sofia/gateway/url/XXXXXXXXXXX
&conference(myConf+flags{moderator,wait-mod})

>From the mod_nibblebill wiki page it seems like we need to set a few
variables for the B leg:

enable_heartbeat_events
nibble_rate
nibble_account=

I was wondering if there is a way to set these variables via the API
call shown above where the conference is created. If not, what options
do you recommend to accomplish this?

Thanks in advance,

Alfredo


From jmesquita at freeswitch.org  Sun Mar 14 11:11:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:11:46 -0300
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>

Try doing this different. Set the effective_caller_id_number, remove the
manual header insertion and bridge it differently.

Something like this:

            $num = "02222222";
            $session->answer();
            $session->setVariable("effective_caller_id_name", "The
Redirector");
            $session->setVariable("effective_caller_id_number",
"0123456789<sip%3A0123456789 at q-loud.net>
");
            $session->execute("bridge", "{sip_from_uri=
sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
}sofia/gateway/mygateway/$num);


/JM


2010/3/13 Roland H?nel <roland at haenel.me>

> Hello,
>
> I'm developing a very simple call-through application using FreeSWITCH, but
> currently
> I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
> header
> for outgoing calls if I want so signal an arbitrary caller ID in the 'From'
> field.
>
> Setup is as follows:
>    - a-leg comes in from an external gateway
>    - dialplan dispatches the call to a perl handler script
>    - perl script looks like this (simplified as much as possible for this
> description):
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The
> Redirector");
>             $session->setVariable("sip_h_P-Asserted-Identity", '<
> sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
>             $session->execute("bridge", "sofia/gateway/mygateway/$num);
>
> The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity
> header as
> set in the perl script. But there is already an existing
> P-Asserted-Identity header by default,
> so I end up with two of them. This is the outgoing INVITE (b-leg):
>
>
> INVITE sip:02222222 at x.x.x.x SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
> Max-Forwards: 63
> From: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
> >;tag=DBSvZ9NZc750g
> To: <sip:02222222 at x.x.x.x>
> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
> CSeq: 128071597 INVITE
> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 244
> X-Port: 5060
> X-FS-Support: update_display
> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
> >
> P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
> >
>
> [... sdp ...]
>
> Numbers are:
>
>     01111111       - the original caller (a-leg & b-leg From:)
>     02222222       - the number redirected to (b-leg To:)
>     0123456789    - the number my provider needs in P-Asserted-Identity
> ('redirector's number')
>
> So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
> line, but unfortunately it already includes another P-Asserted-Identity
> 011111111
> line, and that breaks the setup at my provider.
>
> I really appreciate any help.
>
> Greetings,
> Roland
>
> --
> Roland Haenel
> QSC AG - http://www.qsc.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar 14 11:14:14 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:14:14 -0300
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
Message-ID: <ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>

Or, I can shut-up and let the ppl who know about this stuff reply to your
email.

JM

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> Try doing this different. Set the effective_caller_id_number, remove the
> manual header insertion and bridge it differently.
>
> Something like this:
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The
> Redirector");
>             $session->setVariable("effective_caller_id_number", "
> 0123456789 <sip%3A0123456789 at q-loud.net>");
>             $session->execute("bridge", "{sip_from_uri=
> sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
> }sofia/gateway/mygateway/$num);
>
>
> /JM
>
>
> 2010/3/13 Roland H?nel <roland at haenel.me>
>
>>  Hello,
>>
>> I'm developing a very simple call-through application using FreeSWITCH,
>> but currently
>> I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
>> header
>> for outgoing calls if I want so signal an arbitrary caller ID in the
>> 'From' field.
>>
>> Setup is as follows:
>>    - a-leg comes in from an external gateway
>>    - dialplan dispatches the call to a perl handler script
>>    - perl script looks like this (simplified as much as possible for this
>> description):
>>
>>             $num = "02222222";
>>             $session->answer();
>>             $session->setVariable("effective_caller_id_name", "The
>> Redirector");
>>             $session->setVariable("sip_h_P-Asserted-Identity", '<
>> sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
>>             $session->execute("bridge", "sofia/gateway/mygateway/$num);
>>
>> The problem is now that FreeSWITCH correctly insertes a
>> P-Asserted-Identity header as
>> set in the perl script. But there is already an existing
>> P-Asserted-Identity header by default,
>> so I end up with two of them. This is the outgoing INVITE (b-leg):
>>
>>
>> INVITE sip:02222222 at x.x.x.x SIP/2.0
>> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
>> Max-Forwards: 63
>> From: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>> >;tag=DBSvZ9NZc750g
>> To: <sip:02222222 at x.x.x.x>
>> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
>> CSeq: 128071597 INVITE
>> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, refer
>> Privacy: none
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 244
>> X-Port: 5060
>> X-FS-Support: update_display
>> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>> >
>> P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
>> >
>>
>> [... sdp ...]
>>
>> Numbers are:
>>
>>     01111111       - the original caller (a-leg & b-leg From:)
>>     02222222       - the number redirected to (b-leg To:)
>>     0123456789    - the number my provider needs in P-Asserted-Identity
>> ('redirector's number')
>>
>> So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
>> line, but unfortunately it already includes another P-Asserted-Identity
>> 011111111
>> line, and that breaks the setup at my provider.
>>
>> I really appreciate any help.
>>
>> Greetings,
>> Roland
>>
>> --
>> Roland Haenel
>> QSC AG - http://www.qsc.de
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jmesquita at freeswitch.org  Sun Mar 14 11:15:53 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:15:53 -0300
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
Message-ID: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>

originate {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>
}sofia/gateway/url/XXXXXXXXXXX
&conference(myConf+flags{moderator,wait-mod})


JM


On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> All:
>
> We are currently exploring the possibility of using mod_nibblebill and
> integrating it with a conference. It seems like it offers all the
> features and more that we need. The way we create a conference now is
> as follows:
>
> originate sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> >From the mod_nibblebill wiki page it seems like we need to set a few
> variables for the B leg:
>
> enable_heartbeat_events
> nibble_rate
> nibble_account=
>
> I was wondering if there is a way to set these variables via the API
> call shown above where the conference is created. If not, what options
> do you recommend to accomplish this?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Sun Mar 14 11:17:27 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 11:17:27 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
Message-ID: <354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>

Yes, I'm using the latest SVN.

Spencer

On Mar 14, 2010, at 11:00 AM, Brian West wrote:

> Are you on the latest SVN?
>
> If the problem persists after you update to SVN trunk please read http://wiki.freeswitch.org/wiki/Reporting_Bugs 
>  and report the issue to jira.
>
> /b
>
> On Mar 14, 2010, at 12:49 PM, Spencer Thomason wrote:
>
>> I have Freeswitch running as a SBC in front of several Asterisk  
>> machines and without warning one of the Freeswitch processes will  
>> use 100% of one of the CPU cores and stop responding to SIP  
>> requests.  I'm using the latest svn on Centos 5.4 in a xen vm.   
>> Basically everything will be fine for about 3-4 hours and then this  
>> happens.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Sun Mar 14 11:20:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:20:13 -0500
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
	<ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
Message-ID: <7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>

We are all on like mad today.  :P

/b

On Mar 14, 2010, at 1:14 PM, Jo?o Mesquita wrote:

> Or, I can shut-up and let the ppl who know about this stuff reply to your email.
> 
> JM


From rupa at rupa.com  Sun Mar 14 11:20:45 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 12:20:45 -0600
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
Message-ID: <b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>

Rather than going straight to the conference, you could go through loopback
and have a dialplan that sets the vars and then executes the conference.

On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> All:
>
> We are currently exploring the possibility of using mod_nibblebill and
> integrating it with a conference. It seems like it offers all the
> features and more that we need. The way we create a conference now is
> as follows:
>
> originate sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> >From the mod_nibblebill wiki page it seems like we need to set a few
> variables for the B leg:
>
> enable_heartbeat_events
> nibble_rate
> nibble_account=
>
> I was wondering if there is a way to set these variables via the API
> call shown above where the conference is created. If not, what options
> do you recommend to accomplish this?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From lloyd.aloysius at gmail.com  Sun Mar 14 11:26:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 14:26:50 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
Message-ID: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>

Hi All,

Can some one help me to install Lua .

OS: CentOS 5.4


wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
tar zxf lua-5.1.4.tar.gz
cd lua-5.1.4
make linux

But getting error when running *make linux *
*
*
*Here is the Output*

In file included from lua.h:16,
                 from lua.c:15:
luaconf.h:275:31: error: readline/readline.h: No such file or directory
luaconf.h:276:30: error: readline/history.h: No such file or directory
lua.c: In function ?pushline?:
lua.c:182: warning: implicit declaration of function ?readline?
lua.c:182: warning: assignment makes pointer from integer without a cast
lua.c: In function ?loadline?:
lua.c:210: warning: implicit declaration of function ?add_history?
make[2]: *** [lua.o] Error 1
make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
make: *** [linux] Error 2


Thank you in advance.


Thanks,
Lloyd
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From lawwton at gmail.com  Sun Mar 14 11:26:55 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:26:55 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
Message-ID: <5fe6fa8f1003141126q5380fec5w3f09a5b13115bb72@mail.gmail.com>

Awesome, thanks a lot.

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>:
> originate
> {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>}sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> JM
>
>
> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From yehavi.bourvine at gmail.com  Sun Mar 14 11:33:48 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 14 Mar 2010 20:33:48 +0200
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
	<C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>
Message-ID: <d361ca7b1003141133t5ad5d74ct9f27b99322dc361d@mail.gmail.com>

I am using the almost latest SVN (from yesterday) and our custom dialplan.
I'll try stripping down our dialplan to see whether it helps.

                     Thanks, __Yehavi:

2010/3/14 Brian West <brian at freeswitch.org>

> Are you doing this with the latest SVN and default dialplan or are you
> doing something out of the usual?
>
> /b
>
> On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:
>
> > Hello,
> >
> >    We have a strange behavior which my users told me today, but I have a
> feeling it lasts for a long time (we are running now version 16972).
> >
> > User A calls user B on a Polycom. User B press the "transfer" key and
> calls C who answers. B then press "transfer" again, and now A talks to C.
> However, B is left with a held call to the voicemail of C...
> >
> > Anyone saw this behaviour?  unattended transfer and conference work ok.
> >
> >                              Thanks! __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lawwton at gmail.com  Sun Mar 14 11:38:35 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:38:35 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
Message-ID: <5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>

Thanks Rupa, We have looked at the loopback as an option and really
like the idea, the only thing there is that the wiki page says:

"Please take extra precautions in regard to CDR records. They may not
display the information you are looking for after a call has been
bridged through the loopback endpoint."

We are not quite sure what that statement translates to as far as what
fields from the CDR will be affected, etc...

On Sun, Mar 14, 2010 at 2:20 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Rather than going straight to the conference, you could go through loopback
> and have a dialplan that sets the vars and then executes the conference.
>
> On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nazim.agabekov at gmail.com  Sun Mar 14 11:38:45 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Sun, 14 Mar 2010 22:38:45 +0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
Message-ID: <4B9D2D35.8010303@gmail.com>

Hi Aloysius,

Probably readline development package is not installed. If you have
difficulties compiling Lua,
you could get binaries + includes from:
http://luabinaries.luaforge.net/download.html

On 03/14/2010 10:26 PM, Aloysius Lloyd wrote:
> Hi All,
>
> Can some one help me to install Lua . 
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running *make linux *
> *
> *
> *Here is the Output*
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From rupa at rupa.com  Sun Mar 14 11:40:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 12:40:16 -0600
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
Message-ID: <b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>

That will set it on the a-leg.  IN this case it probably doesn't matter and
it is cleaner to set it on the a-leg.

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> originate {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>
> }sofia/gateway/url/XXXXXXXXXXX
>
> &conference(myConf+flags{moderator,wait-mod})
>
>
> JM
>
>
>
> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil <
> lawwton at gmail.com> wrote:
>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From spencer at 5ninesolutions.com  Sun Mar 14 11:40:17 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 11:40:17 -0700
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
Message-ID: <8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>

I have Centos RPMS and source RPMS for lua 5.1.4 at http://repo.5ninesolutions.com
--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980



On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:

> Hi All,
>
> Can some one help me to install Lua .
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running make linux
>
> Here is the Output
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or  
> directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a  
> cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rm at callrica.co.za  Sun Mar 14 11:41:48 2010
From: rm at callrica.co.za (Roly Maz)
Date: Sun, 14 Mar 2010 20:41:48 +0200
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
Message-ID: <028601cac3a6$056fc940$104f5bc0$@co.za>

Hi Bruce/Brian, 

 

Success on my side too! 

 

Thanks for sharing your experience. I too reset my FS and made minimal
changes to my config. Voila! 

 

Anyone new to FS, and running it on Windows server- I strongly recommend you
pull a copy of Wireshark off the net. It has great tools for analysing
SIP/IP packets. 

 

I'll be adding my config and service provider setup to the wiki this week. 

 

 

Rgds

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 14 March 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

 


Hi Roly,

I promised I'd let you know when I got it working - so here it is:

As Brian said, doublenat is no longer needed.

It looks like I must have just been confusing things horribly by using
various things I'd read about, including:

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

<param name="aggressive-nat-detection" value="true"/>

 

These, and who knows what other mucking about I had been doing, must have
confused everything as none of this is now necessary.

 

All I did - after a reinstall - to get everything working through double
NATs is:

 

1.  In vars.xml 

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>

 

<X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>

 

2.  Then in prefix/sip_profiles/internal.xml

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<param name="ext-sip-ip" value="$${external_sip_ip)"/>

 

Then as the man says, it just works.  

I decided not to enable options pings to keep the far-end NAT open to enable
calling the remote phone - instead I configured it to send keep-alives.

 

And that's it.  I can't believe the knots I was tying myself in.  It just
works !

cheers

Bruce

 

 

 

 

 

 

On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

Ah I see... what? Please share and lead this blind man out the FS
wilderness!

I don't understand...what happens to the external profile? Do you delete it?
And how do you forward port 5060?

...and you thought you were a newbie!

Any insight would be much appreciated...loving the journey.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 09 March 2010 08:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

Ah I see.  I will try again using the internal profile and forwarding
port 5060.  Presumably still creating a directory entry to enable the
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing
>> something wrong here as I am still having difficulty, and still
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to
>> retain an entry in the directory /usr/local/freeeswitch/conf/
>> directory/ext_dns.xml in order to give freeswitch the dns name of
>> the server as a domain for the remote softphone to register on.  I
>> left the group name in this entry the same as inthe default entry,
>> so that the remote phone could register on the same extension
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN
>> as freeswitch) phone to the remote phone, I get the message on the
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment
>> <param name="aggressive-nat-detection" value="true"/> in
>> external.xml, otherwise in the case of a call initiated by the
>> remote phone being hung up by the local phone, freeswitch sent the
>> BYE to the private IP of the remote phone, rather than its public
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org

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


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

 

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From lloyd.aloysius at gmail.com  Sun Mar 14 11:49:27 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 14:49:27 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
Message-ID: <8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>

Nazim & Spencer , thank you.

readline development package install solved the problem. Here is the Lua
install on CentOS 5.4

yum install readline-devel
wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
tar zxf lua-5.1.4.tar.gz
cd lua-5.1.4
make linux
make linux install

Thanks
Lloyd



On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I have Centos RPMS and source RPMS for lua 5.1.4 at
> http://repo.5ninesolutions.com
>  --
> *Spencer Thomason*
> 5Nine Solutions LLC
> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
> p. +1.888.271.7959
> f. +1.310.510.6980
>
>  <http://www.5ninesolutions.com/>
>
> On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>
> Hi All,
>
> Can some one help me to install Lua .
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running *make linux *
> *
> *
> *Here is the Output*
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lawwton at gmail.com  Sun Mar 14 11:50:23 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:50:23 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
	<b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>
Message-ID: <5fe6fa8f1003141150u41d60e71u2d1446d00950efdf@mail.gmail.com>

Thanks Rupa, appreciate the help.

On Sun, Mar 14, 2010 at 2:40 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> That will set it on the a-leg. ?IN this case it probably doesn't matter and
> it is cleaner to set it on the a-leg.
>
> 2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> originate
>> {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>}sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> JM
>>
>>
>> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>>
>>> All:
>>>
>>> We are currently exploring the possibility of using mod_nibblebill and
>>> integrating it with a conference. It seems like it offers all the
>>> features and more that we need. The way we create a conference now is
>>> as follows:
>>>
>>> originate sofia/gateway/url/XXXXXXXXXXX
>>> &conference(myConf+flags{moderator,wait-mod})
>>>
>>> >From the mod_nibblebill wiki page it seems like we need to set a few
>>> variables for the B leg:
>>>
>>> enable_heartbeat_events
>>> nibble_rate
>>> nibble_account=
>>>
>>> I was wondering if there is a way to set these variables via the API
>>> call shown above where the conference is created. If not, what options
>>> do you recommend to accomplish this?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lloyd.aloysius at gmail.com  Sun Mar 14 12:46:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 15:46:46 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
Message-ID: <8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>

I am trying to install Luasql.  Lots of make errors.Here is the steps I
follow

wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
tar xfvz luasql-2.1.1.tar.gz
cd luasql-2.1.1/
vi config

uncomment the mysql
enable mysql and change the mysql lib and include directory path.


How to install LuaSql in CentOS 5.4?

What is the best way to access mysql from Lua?
odbc or natvie?

Thanks
Lloyd


On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Nazim & Spencer , thank you.
>
> readline development package install solved the problem. Here is the Lua
> install on CentOS 5.4
>
> yum install readline-devel
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
> make linux install
>
> Thanks
> Lloyd
>
>
>
> On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I have Centos RPMS and source RPMS for lua 5.1.4 at
>> http://repo.5ninesolutions.com
>>  --
>> *Spencer Thomason*
>> 5Nine Solutions LLC
>> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
>> p. +1.888.271.7959
>> f. +1.310.510.6980
>>
>>  <http://www.5ninesolutions.com/>
>>
>> On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>>
>> Hi All,
>>
>> Can some one help me to install Lua .
>>
>> OS: CentOS 5.4
>>
>>
>> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>> tar zxf lua-5.1.4.tar.gz
>> cd lua-5.1.4
>> make linux
>>
>> But getting error when running *make linux *
>> *
>> *
>> *Here is the Output*
>>
>> In file included from lua.h:16,
>>                  from lua.c:15:
>> luaconf.h:275:31: error: readline/readline.h: No such file or directory
>> luaconf.h:276:30: error: readline/history.h: No such file or directory
>> lua.c: In function ?pushline?:
>> lua.c:182: warning: implicit declaration of function ?readline?
>> lua.c:182: warning: assignment makes pointer from integer without a cast
>> lua.c: In function ?loadline?:
>> lua.c:210: warning: implicit declaration of function ?add_history?
>> make[2]: *** [lua.o] Error 1
>> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>> make[1]: *** [linux] Error 2
>> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>> make: *** [linux] Error 2
>>
>>
>> Thank you in advance.
>>
>>
>> Thanks,
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From rentmycoder at gmail.com  Sun Mar 14 13:01:54 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Sun, 14 Mar 2010 21:01:54 +0100
Subject: [Freeswitch-users] how to access session variables after
	hangup???
Message-ID: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>

> Pass them as arguments to your luarun hangup hook.
And what about variables are not available before the call setup (like
uuid), only during the call progress or after the hangup???
I would need to access the originate_disposition variable...

>> 'luarun hangup.lua ${originate_disposition}'
>> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data (${api_hangup_hook} contains a variable)
any other hint???


> ---------- Forwarded message ----------
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Sun, 14 Mar 2010 13:00:10 -0500
> Subject:?Re: [Freeswitch-users] how to access session variables after hangup???
> Pass them as arguments to your luarun hangup hook.
>
> /b
>
> On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:
>
>> Hi guys,
>> Freeswitch rocks!
>> except the documentation:(
>> I'm not able to figure out how to access session variables after hangup...
>> I try to dial out on a gateway and detect it the called party hangs up
>> before bridge...
>>
>> bgapi originate
>> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
>> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
>>
>> I cannot access the variables using:
>> 1. script arguments:
>> 'luarun hangup.lua ${uuid}'}
>> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
>> (${api_hangup_hook} contains a variable)
>>
>> 2. session:getVariable("originate_disposition")
>> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
>> to call global 'getVar' (a nil value)
>>
>> 3. params:getHeader("variable_sip_req_uri")
>> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
>> to index global 'params' (a nil value)
>>
>> I have asked this before on the dev list too, but Anthony didn't gave
>> me a clean answer here:
>> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
>>
>> "In the script you can access the data from the env event which is
>> automatically created for you and it
>> contains all the channel variables from that departed channel."
>>
>> Ok, but how?????
>>
>> Please help...
>>


From spencer at 5ninesolutions.com  Sun Mar 14 13:13:16 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 13:13:16 -0700
Subject: [Freeswitch-users] Dial number at registered extension
Message-ID: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>

Hello all,
Is there a way I can dial at number at a registered extension.   
Basically what I'd like to do is use Freeswitch as a SBC for a number  
of asterisk machines and I'd like to pass an incoming sip invite to a  
specific number at each one.  Right now I'm specifying the IP in the  
bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if  
I dial the registered user directly it dials the contact address at  
the registered user and does not pass the inbound number.  Basically  
what I'm trying to do is this:

pstn -> ITSP -> freeswitch -> asterisk (1-n)

Thanks,
Spencer


From roland at haenel.me  Sun Mar 14 13:45:12 2010
From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=)
Date: Sun, 14 Mar 2010 21:45:12 +0100
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
	<ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
	<7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>
Message-ID: <429591151003141345n67639652g9fffbafa689c748d@mail.gmail.com>

Thanks guys for being mad like this.
Not only gave me the perfect solution but also some insight about
FreeSWITCH. :-)

Greetings,
Roland
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From nazim.agabekov at gmail.com  Sun Mar 14 13:59:24 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Mon, 15 Mar 2010 00:59:24 +0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
	<8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
Message-ID: <4B9D4E2C.4020407@gmail.com>

I would not use luasql with mysql driver because it leaks memory. Luasql
odbc is better choice for production.
What kind of errors do you have? If you installing luasql.odbc please
make sure you have unix-odbc development packages.

Nazim




On 03/14/2010 11:46 PM, Aloysius Lloyd wrote:
> I am trying to install Luasql.  Lots of make errors.Here is the steps
> I follow
>
> wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
> tar xfvz luasql-2.1.1.tar.gz
> cd luasql-2.1.1/
> vi config
>
> uncomment the mysql
> enable mysql and change the mysql lib and include directory path.
>
>
> How to install LuaSql in CentOS 5.4?
>
> What is the best way to access mysql from Lua? 
> odbc or natvie?
>
> Thanks
> Lloyd
>
>
> On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com <mailto:lloyd.aloysius at gmail.com>> wrote:
>
>     Nazim & Spencer , thank you.
>
>     readline development package install solved the problem. Here is
>     the Lua install on CentOS 5.4
>
>     yum install readline-devel
>     wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>     tar zxf lua-5.1.4.tar.gz
>     cd lua-5.1.4
>     make linux 
>     make linux install
>
>     Thanks
>     Lloyd
>
>
>
>     On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason
>     <spencer at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>>
>     wrote:
>
>         I have Centos RPMS and source RPMS for lua 5.1.4 at
>         http://repo.5ninesolutions.com
>         --
>         *Spencer Thomason*
>         5Nine Solutions LLC
>         e. sales at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>
>         p. +1.888.271.7959
>         f. +1.310.510.6980
>
>         <http://www.5ninesolutions.com/>
>
>         On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>
>>         Hi All,
>>
>>         Can some one help me to install Lua . 
>>
>>         OS: CentOS 5.4
>>
>>
>>         wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>>         tar zxf lua-5.1.4.tar.gz
>>         cd lua-5.1.4
>>         make linux
>>
>>         But getting error when running *make linux *
>>         *
>>         *
>>         *Here is the Output*
>>
>>         In file included from lua.h:16,
>>                          from lua.c:15:
>>         luaconf.h:275:31: error: readline/readline.h: No such file or
>>         directory
>>         luaconf.h:276:30: error: readline/history.h: No such file or
>>         directory
>>         lua.c: In function ?pushline?:
>>         lua.c:182: warning: implicit declaration of function ?readline?
>>         lua.c:182: warning: assignment makes pointer from integer
>>         without a cast
>>         lua.c: In function ?loadline?:
>>         lua.c:210: warning: implicit declaration of function
>>         ?add_history?
>>         make[2]: *** [lua.o] Error 1
>>         make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>>         make[1]: *** [linux] Error 2
>>         make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>>         make: *** [linux] Error 2
>>
>>
>>         Thank you in advance.
>>
>>
>>         Thanks,
>>         Lloyd
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From rupa at rupa.com  Sun Mar 14 14:10:53 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 16:10:53 -0500
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
	<5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>
Message-ID: <b39273531003141410q12a580b3s8c8dd781eb3f66d7@mail.gmail.com>

Basically you'll get new CDR records for the loopback side of the call. So
you may have to reconcile the two -- depends on how you process your CDRs.
 If you can avoid loopback, do so.  In this case, doing nibblebill on the
a-leg is the right way to go unless you have a specific reason to nibble on
the b-leg.

On Sun, Mar 14, 2010 at 1:38 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Thanks Rupa, We have looked at the loopback as an option and really
> like the idea, the only thing there is that the wiki page says:
>
> "Please take extra precautions in regard to CDR records. They may not
> display the information you are looking for after a call has been
> bridged through the loopback endpoint."
>
> We are not quite sure what that statement translates to as far as what
> fields from the CDR will be affected, etc...
>
> On Sun, Mar 14, 2010 at 2:20 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> > Rather than going straight to the conference, you could go through
> loopback
> > and have a dialplan that sets the vars and then executes the conference.
> >
> > On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> All:
> >>
> >> We are currently exploring the possibility of using mod_nibblebill and
> >> integrating it with a conference. It seems like it offers all the
> >> features and more that we need. The way we create a conference now is
> >> as follows:
> >>
> >> originate sofia/gateway/url/XXXXXXXXXXX
> >> &conference(myConf+flags{moderator,wait-mod})
> >>
> >> >From the mod_nibblebill wiki page it seems like we need to set a few
> >> variables for the B leg:
> >>
> >> enable_heartbeat_events
> >> nibble_rate
> >> nibble_account=
> >>
> >> I was wondering if there is a way to set these variables via the API
> >> call shown above where the conference is created. If not, what options
> >> do you recommend to accomplish this?
> >>
> >> Thanks in advance,
> >>
> >> Alfredo
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Sun Mar 14 14:12:28 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 16:12:28 -0500
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
Message-ID: <b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>

maybe look at using something like mod_easyroute.  You setup a database of
mappings between DID and endpoints.  It is designed for the situation you
describe.

On Sun, Mar 14, 2010 at 3:13 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> Is there a way I can dial at number at a registered extension.
> Basically what I'd like to do is use Freeswitch as a SBC for a number
> of asterisk machines and I'd like to pass an incoming sip invite to a
> specific number at each one.  Right now I'm specifying the IP in the
> bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if
> I dial the registered user directly it dials the contact address at
> the registered user and does not pass the inbound number.  Basically
> what I'm trying to do is this:
>
> pstn -> ITSP -> freeswitch -> asterisk (1-n)
>
> Thanks,
> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From spencer at 5ninesolutions.com  Sun Mar 14 15:02:33 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 15:02:33 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
Message-ID: <AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>

That looks promising, thanks.  I was hoping to have FS do it  
automatically.  I.e. inbound dids are assigned to specific asterisk  
boxes, ITSP dials inbound did of freeswitch, freeswitch dials inbound  
did @ whereever the registration came from instead of having the IP  
address hard coded into a route.


On Mar 14, 2010, at 2:12 PM, Rupa Schomaker wrote:

> maybe look at using something like mod_easyroute.  You setup a  
> database of mappings between DID and endpoints.  It is designed for  
> the situation you describe.
>
> On Sun, Mar 14, 2010 at 3:13 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> Hello all,
> Is there a way I can dial at number at a registered extension.
> Basically what I'd like to do is use Freeswitch as a SBC for a number
> of asterisk machines and I'd like to pass an incoming sip invite to a
> specific number at each one.  Right now I'm specifying the IP in the
> bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if
> I dial the registered user directly it dials the contact address at
> the registered user and does not pass the inbound number.  Basically
> what I'm trying to do is this:
>
> pstn -> ITSP -> freeswitch -> asterisk (1-n)
>
> Thanks,
> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Mar 14 15:18:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 17:18:05 -0500
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
Message-ID: <2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>

You sure can still have it call a registration.. You're mixing the dialplan and user directly as if they are a single entity.  They aren't.

/b

On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:

> That looks promising, thanks.  I was hoping to have FS do it automatically.  I.e. inbound dids are assigned to specific asterisk boxes, ITSP dials inbound did of freeswitch, freeswitch dials inbound did @ whereever the registration came from instead of having the IP address hard coded into a route.



From spencer at 5ninesolutions.com  Sun Mar 14 16:00:26 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 16:00:26 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
Message-ID: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>

I guess my question is, is there some function which will return the  
IP address of a registered user?  Similar to sofia_contact but only  
the IP address?

Spencer

On Mar 14, 2010, at 3:18 PM, Brian West wrote:

> You sure can still have it call a registration.. You're mixing the  
> dialplan and user directly as if they are a single entity.  They  
> aren't.
>
> /b
>
> On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>
>> That looks promising, thanks.  I was hoping to have FS do it  
>> automatically.  I.e. inbound dids are assigned to specific asterisk  
>> boxes, ITSP dials inbound did of freeswitch, freeswitch dials  
>> inbound did @ whereever the registration came from instead of  
>> having the IP address hard coded into a route.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Sun Mar 14 16:08:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Mar 2010 17:08:47 -0600
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
Message-ID: <191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>

Do you think its due to resource limitations in your vm?
Make sure you check all the config options to give fs full access to all the
resouces.

If it persists, do top -H and identify the pid with the 100% usage.  Then
get a gcore and gdb it to capture the output of "thread apply all bt"

Report the findings to jira.

On Mar 14, 2010 1:23 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
wrote:

Yes, I'm using the latest SVN.

Spencer


On Mar 14, 2010, at 11:00 AM, Brian West wrote:

> Are you on the latest SVN?
>
> If the problem pe...
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From jmesquita at freeswitch.org  Sun Mar 14 16:17:13 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 20:17:13 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
Message-ID: <ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>

Yes, sofia_contact with regex to extract the IP portion? You had the answer
yourself.

Jo?o Mesquita



On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sun Mar 14 16:17:36 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 20:17:36 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
Message-ID: <ebbc2d551003141617n12b2f81djfba1e6d6a7b8fe66@mail.gmail.com>

Oh, check the RURI example on default configs. That might give you a hint.

Jo?o Mesquita



On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Sun Mar 14 17:14:11 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:14:11 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
Message-ID: <E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>

No.  Its on our hardware, we just use xen to virtualize the machines..  
this is the only vm on the machine.  Gcore complains about no  
debugging symbols found, do I need to rebuild?

It appears to be a problem when registering to one of my trunks.  The  
very strange thing is the everything will be working perfectly and  
then all of a sudden this happens, i.e. trunks regged, calls  
completing, etc. This is a very lightly loaded server.

This is the output from the console:

freeswitch at internal> 2010-03-14 23:12:15.199983 [NOTICE] sofia_reg.c: 
336 Registering bandtel
2010-03-14 23:12:15.223977 [DEBUG] sofia_reg.c:1478 Changing expire  
time to 475 by request of proxy sip:66.237.65.67
2010-03-14 23:20:05.429613 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:20:08.285756 [ERR] sofia_reg.c:1490 bandtel Registration  
Failed with status Operation has no matching challenge  [904]. failure  
#1
2010-03-14 23:20:10.043933 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:21:16.332797 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:21:52.528409 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:22:56.945511 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:23:33.083177 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:24:37.398155 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:25:13.631825 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:26:17.978767 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:26:54.172425 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:27:58.579354 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:28:34.605017 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.


On Mar 14, 2010, at 4:08 PM, Anthony Minessale wrote:

> Do you think its due to resource limitations in your vm?
> Make sure you check all the config options to give fs full access to  
> all the resouces.
>
> If it persists, do top -H and identify the pid with the 100% usage.   
> Then get a gcore and gdb it to capture the output of "thread apply  
> all bt"
>
> Report the findings to jira.
>
>> On Mar 14, 2010 1:23 PM, "Spencer Thomason" <spencer at 5ninesolutions.com 
>> > wrote:
>>
>> Yes, I'm using the latest SVN.
>>
>> Spencer
>>
>> On Mar 14, 2010, at 11:00 AM, Brian West wrote:
>>
>> > Are you on the latest SVN?
>> >
>> > If the problem pe...
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Mar 14 17:23:43 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 19:23:43 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
Message-ID: <F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>

Are you sure you're on SVN trunk?

/b

On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:

> No.  Its on our hardware, we just use xen to virtualize the machines.. this is the only vm on the machine.  Gcore complains about no debugging symbols found, do I need to rebuild?  
> 
> It appears to be a problem when registering to one of my trunks.  The very strange thing is the everything will be working perfectly and then all of a sudden this happens, i.e. trunks regged, calls completing, etc. This is a very lightly loaded server.
> 
> This is the output from the console:



From spencer at 5ninesolutions.com  Sun Mar 14 17:25:43 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:25:43 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
Message-ID: <10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>

Duh.. :-)  Thanks!

On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:

> Yes, sofia_contact with regex to extract the IP portion? You had the  
> answer yourself.
>
> Jo?o Mesquita
>
>
>
> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From spencer at 5ninesolutions.com  Sun Mar 14 17:38:43 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:38:43 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
Message-ID: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>

I rebuilt today.

On Mar 14, 2010, at 5:23 PM, Brian West wrote:

> Are you sure you're on SVN trunk?
>
> /b
>
> On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
>
>> No.  Its on our hardware, we just use xen to virtualize the  
>> machines.. this is the only vm on the machine.  Gcore complains  
>> about no debugging symbols found, do I need to rebuild?
>>
>> It appears to be a problem when registering to one of my trunks.   
>> The very strange thing is the everything will be working perfectly  
>> and then all of a sudden this happens, i.e. trunks regged, calls  
>> completing, etc. This is a very lightly loaded server.
>>
>> This is the output from the console:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Sun Mar 14 17:44:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 19:44:36 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
Message-ID: <C6CA5510-F725-4612-B39C-AF0E959A1E3E@freeswitch.org>

Collect sip traces for that issue.

/b

On Mar 14, 2010, at 7:38 PM, Spencer Thomason wrote:

> I rebuilt today.
> 
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
> 
>> Are you sure you're on SVN trunk?
>> 
>> /b



From jbrucehopkins at gmail.com  Sun Mar 14 18:53:10 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 15 Mar 2010 01:53:10 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <028601cac3a6$056fc940$104f5bc0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
Message-ID: <d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>

Doubles all round !

cheers
Bruce

On 14 March 2010 18:41, Roly Maz <rm at callrica.co.za> wrote:

>  Hi Bruce/Brian,
>
>
>
> Success on my side too!
>
>
>
> Thanks for sharing your experience. I too reset my FS and made minimal
> changes to my config. Voila!
>
>
>
> Anyone new to FS, and running it on Windows server? I strongly recommend
> you pull a copy of Wireshark off the net. It has great tools for analysing
> SIP/IP packets.
>
>
>
> I?ll be adding my config and service provider setup to the wiki this week.
>
>
>
>
>
> Rgds
>
>
>
>
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bruce
> Hopkins
> *Sent:* 14 March 2010 12:57 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] doublenat outgoing call issue
>
>
>
>
> Hi Roly,
>
> I promised I'd let you know when I got it working - so here it is:
>
> As Brian said, doublenat is no longer needed.
>
> It looks like I must have just been confusing things horribly by using
> various things I'd read about, including:
>
> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
>
> <param name="aggressive-nat-detection" value="true"/>
>
>
>
> These, and who knows what other mucking about I had been doing, must have
> confused everything as none of this is now necessary.
>
>
>
> All I did - after a reinstall - to get everything working through double
> NATs is:
>
>
>
> 1.  In vars.xml
>
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>
>
>
>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>
>
>
>
> 2.  Then in prefix/sip_profiles/internal.xml
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_sip_ip)"/>
>
>
>
> Then as the man says, it just works.
>
> I decided not to enable options pings to keep the far-end NAT open to
> enable calling the remote phone - instead I configured it to send
> keep-alives.
>
>
>
> And that's it.  I can't believe the knots I was tying myself in.  It just
> works !
>
> cheers
>
> Bruce
>
>
>
>
>
>
>
>
>
>
>
>
>
> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>
> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Mar 14 19:06:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 21:06:15 -0500
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
Message-ID: <A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>

Its nice to hear more people have this working.  The more that depend on this .. the simpler my job is... because If we break it they start to open jira's about it.

/b

On Mar 14, 2010, at 8:53 PM, Bruce Hopkins wrote:

> Doubles all round !
> 
> cheers
> Bruce



From jason at jasonjgw.net  Sun Mar 14 19:17:46 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Mar 2010 13:17:46 +1100
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
References: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
	<A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
Message-ID: <20100315021746.GA31211@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Its nice to hear more people have this working.  The more that depend on
> this .. the simpler my job is... because If we break it they start to open
> jira's about it.

If both parties behind NAT could just go and register with
http://tunelbroker.net/ or http://www.freenet6.net/ or http://www.sixxs.net/
or a similar service, or lobby their ISPs to introduce native IPv6, this
problem would start to go away.



From spencer at 5ninesolutions.com  Sun Mar 14 19:20:50 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 19:20:50 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
Message-ID: <3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>

So just in case anyone else ever needs to do this.. here's what I  
ended up with:

<include>
   <extension name="NNNNNNNNNNN-did">
     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="set" data="accountcode=<ACCTNUM>"/>
       <action application="bridge" data="sofia/default/$1${regex($ 
{sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}" />
     </condition>
   </extension>
</include>

Basically the extension UAs register to is created as the same as the  
account code.  We then use their ip at registration to dial a did on  
their end.

Spencer


On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:

> Duh.. :-)  Thanks!
>
> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>
>> Yes, sofia_contact with regex to extract the IP portion? You had  
>> the answer yourself.
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <spencer at 5ninesolutions.com 
>> > wrote:
>> I guess my question is, is there some function which will return the
>> IP address of a registered user?  Similar to sofia_contact but only
>> the IP address?
>>
>> Spencer
>>
>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>
>> > You sure can still have it call a registration.. You're mixing the
>> > dialplan and user directly as if they are a single entity.  They
>> > aren't.
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>> >
>> >> That looks promising, thanks.  I was hoping to have FS do it
>> >> automatically.  I.e. inbound dids are assigned to specific  
>> asterisk
>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>> >> inbound did @ whereever the registration came from instead of
>> >> having the IP address hard coded into a route.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Sun Mar 14 19:40:27 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 23:40:27 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
	<3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
Message-ID: <ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>

Wiki is your friend to that type of info! Contribute to it, please!

JM


On Sun, Mar 14, 2010 at 11:20 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> So just in case anyone else ever needs to do this.. here's what I ended up
> with:
>
> <include>
>   <extension name="NNNNNNNNNNN-did">
>     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="set" data="accountcode=<ACCTNUM>"/>
>       <action application="bridge"
> data="sofia/default/$1${regex(${sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}"
> />
>     </condition>
>   </extension>
> </include>
>
> Basically the extension UAs register to is created as the same as the
> account code.  We then use their ip at registration to dial a did on their
> end.
>
> Spencer
>
>
> On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:
>
> Duh.. :-)  Thanks!
>
> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>
> Yes, sofia_contact with regex to extract the IP portion? You had the answer
> yourself.
>
> Jo?o Mesquita
>
>
>
> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I guess my question is, is there some function which will return the
>> IP address of a registered user?  Similar to sofia_contact but only
>> the IP address?
>>
>> Spencer
>>
>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>
>> > You sure can still have it call a registration.. You're mixing the
>> > dialplan and user directly as if they are a single entity.  They
>> > aren't.
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>> >
>> >> That looks promising, thanks.  I was hoping to have FS do it
>> >> automatically.  I.e. inbound dids are assigned to specific asterisk
>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>> >> inbound did @ whereever the registration came from instead of
>> >> having the IP address hard coded into a route.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Sun Mar 14 21:36:59 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 00:36:59 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <4B9D4E2C.4020407@gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
	<8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
	<4B9D4E2C.4020407@gmail.com>
Message-ID: <8a19bf2e1003142136x17dbddd2j2c745b5265cd5a92@mail.gmail.com>

Thank you for the suggestion. Now FreeSWITCH , Lua , LuaSQL (ODBC)
working great.


On Sun, Mar 14, 2010 at 4:59 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  I would not use luasql with mysql driver because it leaks memory. Luasql
> odbc is better choice for production.
> What kind of errors do you have? If you installing luasql.odbc please make
> sure you have unix-odbc development packages.
>
> Nazim
>
>
>
>
>
> On 03/14/2010 11:46 PM, Aloysius Lloyd wrote:
>
> I am trying to install Luasql.  Lots of make errors.Here is the steps I
> follow
>
>  wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
> tar xfvz luasql-2.1.1.tar.gz
> cd luasql-2.1.1/
> vi config
>
>  uncomment the mysql
> enable mysql and change the mysql lib and include directory path.
>
>
>  How to install LuaSql in CentOS 5.4?
>
>  What is the best way to access mysql from Lua?
> odbc or natvie?
>
>  Thanks
> Lloyd
>
>
> On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Nazim & Spencer , thank you.
>>
>>  readline development package install solved the problem. Here is the Lua
>> install on CentOS 5.4
>>
>>  yum install readline-devel
>>  wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>> tar zxf lua-5.1.4.tar.gz
>> cd lua-5.1.4
>> make linux
>>  make linux install
>>
>>  Thanks
>> Lloyd
>>
>>
>>
>> On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
>> spencer at 5ninesolutions.com> wrote:
>>
>>> I have Centos RPMS and source RPMS for lua 5.1.4 at
>>> http://repo.5ninesolutions.com
>>>   --
>>> *Spencer Thomason*
>>> 5Nine Solutions LLC
>>> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
>>> p. +1.888.271.7959
>>> f. +1.310.510.6980
>>>
>>>  <http://www.5ninesolutions.com/>
>>>
>>>   On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>>>
>>>   Hi All,
>>>
>>>  Can some one help me to install Lua .
>>>
>>>  OS: CentOS 5.4
>>>
>>>
>>>  wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>>> tar zxf lua-5.1.4.tar.gz
>>> cd lua-5.1.4
>>> make linux
>>>
>>>  But getting error when running *make linux *
>>> *
>>> *
>>> *Here is the Output*
>>>
>>>  In file included from lua.h:16,
>>>                  from lua.c:15:
>>> luaconf.h:275:31: error: readline/readline.h: No such file or directory
>>> luaconf.h:276:30: error: readline/history.h: No such file or directory
>>> lua.c: In function ?pushline?:
>>> lua.c:182: warning: implicit declaration of function ?readline?
>>> lua.c:182: warning: assignment makes pointer from integer without a cast
>>> lua.c: In function ?loadline?:
>>> lua.c:210: warning: implicit declaration of function ?add_history?
>>> make[2]: *** [lua.o] Error 1
>>> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>>> make[1]: *** [linux] Error 2
>>> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>>> make: *** [linux] Error 2
>>>
>>>
>>>  Thank you in advance.
>>>
>>>
>>>  Thanks,
>>> Lloyd
>>>   _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tzury.by at reguluslabs.com  Sun Mar 14 23:52:39 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 08:52:39 +0200
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
Message-ID: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>

Hi,

Owe to an hardware failure I have had to reinstall my server.
Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap

Getting recommendation to try the new sangoma prid I tried to follow the
instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install

Being unable to make it work, I search through the mailing-list and saw the
specific versions which should be used.
Followed these instruction as well, yet, it is not working.

probing hardware shows all goos indications.

When I try to originate from within the FreeSWITCH's CLI I get

freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML public
2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
online.

-ERR NORMAL_CIRCUIT_CONGESTION

2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
freeswitch at snaptrunk-02>


when I try to run #smg_ctrl start
I get:

smg_ctrl: sangoma_prid failed to start
> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>

*However, that file ( /var/log/sangoma_mgd.log) is empty.

*I pastebin all relevant files at http://gist.github.com/332589*

*I am using the latest freeswitch (svn trunk)

I tried the smg with these versions (from
ftp://ftp.sangoma.com/linux/custom/DavidYS/)

   - wanpipe-3.5.10.smg_pri.1.tgz
   - wanpipe-3.5.10.3.smg_pri-v1.58.tgz

*

*
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From gmaruzz at celliax.org  Mon Mar 15 01:15:55 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Mar 2010 09:15:55 +0100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
Message-ID: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>

Ciao FreeSWITCHers,

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

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

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

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

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jason at jasonjgw.net  Mon Mar 15 01:31:56 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Mar 2010 19:31:56 +1100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
In-Reply-To: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>
References: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>
Message-ID: <20100315083156.GA12318@jdc.jasonjgw.net>

Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Ciao FreeSWITCHers,
> 
> is with pleasure that we're announcing the renaming of mod_skypiax
> module to mod_skypopen.

Could you also rename it in the Debian packaging files (located in the debian
directory)? 



From gmaruzz at celliax.org  Mon Mar 15 01:47:56 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Mar 2010 09:47:56 +0100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
In-Reply-To: <20100315083156.GA12318@jdc.jasonjgw.net>
References: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com> 
	<20100315083156.GA12318@jdc.jasonjgw.net>
Message-ID: <7b197bef1003150147t7837e332jc2497197eaf2bc77@mail.gmail.com>

On Mon, Mar 15, 2010 at 9:31 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Could you also rename it in the Debian packaging files (located in the debian
> directory)?

I'll ask the people who's taking care of Debian packages (I'll find
out who) to do it, thanks for reporting.

-giovanni

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



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From maciej.aniserowicz at gmail.com  Mon Mar 15 01:56:36 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 15 Mar 2010 00:56:36 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL cause
Message-ID: <1268643396699-4735736.post@n2.nabble.com>


Hello,
I have the following problem:

I register with x-lite to my freeswitch and then call it by sending command
bgapi originate
{originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
&park()

Then i dial through a gateway which points to another freeswitch instance:
bgapi originate
{originate_timeout=20}[24cc_session_id=3134a5ae-ec00-4164-94b1-a7fb5e0368fa,ignore_early_media=true,max_forwards=999999,origination_uuid=a85fd47e-051f-42ea-987e-f4e090c7ae78]sofia/gateway/prospect_gateway/1300671139878747511456
&park()

The dialplan causes answering the phone, playing a file and hanging up.

The two calls are connected with this command:

bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline


Unfortunately after the second call hangs up, the first call gets hanged up
too.

I recently upgraded to FS trunk.

Logs are in pastebin:
first freeswitch: http://pastebin.freeswitch.org/12432
second freeswitch: http://pastebin.freeswitch.org/12433

Please take a look and help me figure this out..
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4735736.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Mon Mar 15 02:20:03 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Mar 2010 10:20:03 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>

I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson

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From dujinfang at gmail.com  Mon Mar 15 03:08:30 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 15 Mar 2010 18:08:30 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
Message-ID: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>

ALL,

I just upgrade FS to 16992.

1) timer warning: How's possible the 10000 microseconds? audio sounds ok.


[WARNING] switch_time.c:206 Timer resolution of 10000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may
experience audio problems.

uname -a
Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
2008 i686 GNU/Linux


I used a script to check the interupts

:~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
LOC:   14892854   14892843   14892841   14892830
LOC:   14892955   14892944   14892942   14892931
LOC:   14893056   14893045   14893043   14893032
LOC:   14893157   14893146   14893144   14893133
LOC:   14893258   14893247   14893245   14893234
LOC:   14893359   14893348   14893346   14893335
LOC:   14893460   14893449   14893447   14893436


2) run dingaling with client mode. dingaling doesn't pick up answer. I
originate a call from FS, the console blocks, gtalk client ring and it
looks ok when I hit answer, but no sound, FS console still blocks. a
few seconds later gtalk client shows:

Sorry! The voice chat with language_lesson failed because of a network
problem at 5:50 PM. Please try again.
 Click here to report this to Google.


here is a log with bgapi:
http://pastebin.freeswitch.org/12434


rev 14696 works on the same server same conf. another server running
16958M works at the same time. I have no clue to check, is it related
to the timer as in 1)? can some help me to take a look? let me know if
you need "dl_debug on" logs.

Thanks.


From dujinfang at gmail.com  Mon Mar 15 03:27:38 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 15 Mar 2010 18:27:38 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
Message-ID: <23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>

here is the client:

<profile type="client">
 <param name="name" value="gmail.com"/>
 <param name="login" value="xx at gmail.com/talk"/>
 <param name="password" value="password"/>
 <param name="server" value="talk.google.com"/>
 <param name="dialplan" value="XML"/>
 <param name="contex" value="public"/>
 <param name="message" value="FreeSWITCH"/>
 <param name="rtp-ip" value="auto"/>
 <!-- if you are behind NAT, you should probably use stun -->
 <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
 <param name="auto-login" value="true"/>
 <param name="auto-reply" value="Welcome"/>
 <param name="sasl" value="plain"/>
 <param name="tls" value="true"/>
 <param name="exten" value="5000"/>
<!-- <param name="local-network-acl" value="localnet.auto"/> -->
</profile>

2010/3/15 Seven Du <dujinfang at gmail.com>:
> ALL,
>
> I just upgrade FS to 16992.
>
> 1) timer warning: How's possible the 10000 microseconds? audio sounds ok.
>
>
> [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds detected!
> Do you have your kernel timer set to higher than 1 kHz? You may
> experience audio problems.
>
> uname -a
> Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> 2008 i686 GNU/Linux
>
>
> I used a script to check the interupts
>
> :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
> LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
> LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
> LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
> LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
> LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
> LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>
>
> 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> originate a call from FS, the console blocks, gtalk client ring and it
> looks ok when I hit answer, but no sound, FS console still blocks. a
> few seconds later gtalk client shows:
>
> Sorry! The voice chat with language_lesson failed because of a network
> problem at 5:50 PM. Please try again.
> ?Click here to report this to Google.
>
>
> here is a log with bgapi:
> http://pastebin.freeswitch.org/12434
>
>
> rev 14696 works on the same server same conf. another server running
> 16958M works at the same time. I have no clue to check, is it related
> to the timer as in 1)? can some help me to take a look? let me know if
> you need "dl_debug on" logs.
>
> Thanks.
>


From tzury.by at reguluslabs.com  Mon Mar 15 03:28:08 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 12:28:08 +0200
Subject: [Freeswitch-users] regarding the recent sangoma boost
Message-ID: <10128ef11003150328t3e033993nb4de567d9e129bd6@mail.gmail.com>

one must run # mkdir /root/sangoma before head.
otherwise, sangoma_prid will fail to start


I discoverd this since /var/log/sangoma_mgd.log file was empty even though
the line
local2.*    /var/log/sangoma_mgd.log
exists in /etc/syslog.conf

I did the following:
# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# /usr/sbin/sangoma_prid &> sangoma_prid.log

an then with # tail -f sangoma_prid.log
I got
==> sangoma_prid.log <==
================System restart=============
= Sangoma PRI Protocol Stack Daemon       =
= Version: 1.58                           =
= Date: Feb 18 2010                       =
= Wanpipe Release: wanpipe-3.5.10.3          =
= Revision:Revision: 15364          =
===========================================
sh: gen_fs_gcore.sh: not found
touch: cannot touch `/root/sangoma/sangoma_prid.gcore.24821': No such file
or directory
/usr/bin/gcore: 83: cannot create /root/sangoma/sangoma_prid.gcore.24821:
Directory nonexistent
gcore: failed to create /root/sangoma/sangoma_prid.gcore.24794


I then ran `# mkdir /root/sangoma` and restart everything

Right now I am facing another problem which hope to over come right after
lunch

the problem is this error

freeswitch at snaptrunk-02> originate openzap/1/a/0525133399 XML public
2010-03-15 12:20:56.307066 [WARNING] ozmod_sangoma_boost.c:341 TX EVENT:
CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[N/A] Cd=[0525133399] Ci=[N/A]
Rdnis=[]



2010-03-15 12:21:06.858887 [WARNING] sangoma_boost_client.c:221 TX EVENT
(N): CALL_START_NACK:(82) [w1g1] Rc=0 CSid=1 Seq=1
2010-03-15 12:21:06.858887 [ERR] switch_ivr_originate.c:2422 Cannot create
outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE]

-ERR RECOVERY_ON_TIMER_EXPIRE

2010-03-15 12:21:06.858887 [DEBUG] switch_ivr_originate.c:3220 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
freeswitch at snaptrunk-02>
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From jbrucehopkins at gmail.com  Mon Mar 15 04:39:26 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 15 Mar 2010 11:39:26 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <20100315021746.GA31211@jdc.jasonjgw.net>
References: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
	<A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
	<20100315021746.GA31211@jdc.jasonjgw.net>
Message-ID: <d03a10e01003150439q334b88d9t5f18f97bc6e4fcee@mail.gmail.com>

Very interesting.  Thanks for sharing this.  I must read up on it.

cheersBruce

On 15 March 2010 02:17, Jason White <jason at jasonjgw.net> wrote:

> Brian West <brian at freeswitch.org> wrote:
> > Its nice to hear more people have this working.  The more that depend on
> > this .. the simpler my job is... because If we break it they start to
> open
> > jira's about it.
>
> If both parties behind NAT could just go and register with
> http://tunelbroker.net/ or http://www.freenet6.net/ or
> http://www.sixxs.net/
> or a similar service, or lobby their ISPs to introduce native IPv6, this
> problem would start to go away.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lakindia89 at gmail.com  Mon Mar 15 05:38:56 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 15 Mar 2010 18:08:56 +0530
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
In-Reply-To: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
References: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
Message-ID: <7d79b3931003150538u4621c056l2469f77fd78d0ad7@mail.gmail.com>

Make sure that the following packages are installed.

apt-get install libsctp1 patch bison build-essential gcc g++ make
libncurses5-dev flex linux-headers-$(uname -r) automake libtools autoconf
libtermcap-devel devscripts pbuilder glibc2.4



On Mon, Mar 15, 2010 at 12:22 PM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

> Hi,
>
> Owe to an hardware failure I have had to reinstall my server.
> Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap
>
> Getting recommendation to try the new sangoma prid I tried to follow the
> instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> Being unable to make it work, I search through the mailing-list and saw the
> specific versions which should be used.
> Followed these instruction as well, yet, it is not working.
>
> probing hardware shows all goos indications.
>
> When I try to originate from within the FreeSWITCH's CLI I get
>
> freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML
> public
> 2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
> online.
>
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> freeswitch at snaptrunk-02>
>
>
> when I try to run #smg_ctrl start
> I get:
>
> smg_ctrl: sangoma_prid failed to start
>> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>>
>
> *However, that file ( /var/log/sangoma_mgd.log) is empty.
>
> *I pastebin all relevant files at http://gist.github.com/332589*
>
> *I am using the latest freeswitch (svn trunk)
>
> I tried the smg with these versions (from
> ftp://ftp.sangoma.com/linux/custom/DavidYS/)
>
>    - wanpipe-3.5.10.smg_pri.1.tgz
>    - wanpipe-3.5.10.3.smg_pri-v1.58.tgz
>
> *
>
> *
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Mon Mar 15 06:40:25 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 09:40:25 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
Message-ID: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>

Hi All,

I have the following simple Lua script answer a inbound call then dial
another number and bridge the call.

Is there any way to get the  call duration or billing sec using the

session:getVariable(  );



-- answer the call
session:answer();

-- sleep a second
session:sleep(1000);

-- Initiate an outbound call
obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)

obSession:execute("sched_hangup","+60 alloted_timeout");

-- bride the call
    freeswitch.bridge(session, obSession);

-- hangup
session:hangup();


Thank you
Lloyd
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From tzury.by at reguluslabs.com  Mon Mar 15 06:45:51 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 15:45:51 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error using
	Sangoma (sangoma_prid)
Message-ID: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>

Hi,

This is my setup

FreeSWITCH: Version 1.0.trunk (16992M)
Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58

I have finally managed to set up the environment coping many hick-ups
reported in this mailing list recently.
However, I am still encountering problems when trying to perform call from
SIP to PRI.

I have configured my dialplan to bridge all numbers which are not

pastebin for all configuration files is available at:
http://pastebin.freeswitch.org/12437

pastebin for CLI output is available at:
http://pastebin.freeswitch.org/12436
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From tzury.by at reguluslabs.com  Mon Mar 15 06:51:51 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 15:51:51 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
Message-ID: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>

Hi,

This is my setup

FreeSWITCH: Version 1.0.trunk (16992M)
Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58

I have finally managed to set up the environment coping many hick-ups
reported in this mailing list recently.
However, I am still encountering problems when trying to perform call
from SIP to PRI.

I have configured my dialplan to bridge all numbers which are not SIP
to be bridged over to the PSTN.
However, when calling a PSTN number I am getting
RECOVERY_ON_TIMER_EXPIRE error

pastebin for all configuration files is available at:
http://pastebin.freeswitch.org/12437

pastebin for CLI output is available at:
http://pastebin.freeswitch.org/12436


From sos at sokhapkin.dyndns.org  Mon Mar 15 06:56:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 09:56:37 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
Message-ID: <201003150956.37356.sos@sokhapkin.dyndns.org>

Unfortunately all "interesting" billing-related channel variables are set 
after the lua script execution, when channel enters REPORTING state. To access 
those variables I execute separate lua script using mod_cdr_csv, it allows to 
invoke lua script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> Hi All,
> 
> I have the following simple Lua script answer a inbound call then dial
> another number and bridge the call.
> 
> Is there any way to get the  call duration or billing sec using the
> 
> session:getVariable(  );
> 
> 
> 
> -- answer the call
> session:answer();
> 
> -- sleep a second
> session:sleep(1000);
> 
> -- Initiate an outbound call
> obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> 
> obSession:execute("sched_hangup","+60 alloted_timeout");
> 
> -- bride the call
>     freeswitch.bridge(session, obSession);
> 
> -- hangup
> session:hangup();
> 
> 
> Thank you
> Lloyd
> 



From moises.silva at gmail.com  Mon Mar 15 07:00:20 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 10:00:20 -0400
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
In-Reply-To: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
References: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
Message-ID: <c4d05cbe1003150700k455a1a5ag82668bee64d19f5c@mail.gmail.com>

Enable debug and do svn update. I suspect you don't have sctp.

You can also run sangoma_prid directly to see the output in stdout/stderr

# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# sangoma_prid

If there is no output in /var/log may be is because /etc/syslog.conf is
missing something like this:

# Sangoma Media Gateway log
local2.*                /var/log/sangoma_mgd.log


# Sangoma BRI Daemon (smg_bri)  log
local3.*                /var/log/sangoma_bri.log


On Mon, Mar 15, 2010 at 2:52 AM, Tzury Bar Yochay
<tzury.by at reguluslabs.com>wrote:

> Hi,
>
> Owe to an hardware failure I have had to reinstall my server.
> Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap
>
> Getting recommendation to try the new sangoma prid I tried to follow the
> instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> Being unable to make it work, I search through the mailing-list and saw the
> specific versions which should be used.
> Followed these instruction as well, yet, it is not working.
>
> probing hardware shows all goos indications.
>
> When I try to originate from within the FreeSWITCH's CLI I get
>
> freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML
> public
> 2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
> online.
>
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> freeswitch at snaptrunk-02>
>
>
> when I try to run #smg_ctrl start
> I get:
>
> smg_ctrl: sangoma_prid failed to start
>> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>>
>
> *However, that file ( /var/log/sangoma_mgd.log) is empty.
>
> *I pastebin all relevant files at http://gist.github.com/332589*
>
> *I am using the latest freeswitch (svn trunk)
>
> I tried the smg with these versions (from
> ftp://ftp.sangoma.com/linux/custom/DavidYS/)
>
>    - wanpipe-3.5.10.smg_pri.1.tgz
>    - wanpipe-3.5.10.3.smg_pri-v1.58.tgz
>
> *
>
> *
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Mon Mar 15 07:03:22 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 10:03:22 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
Message-ID: <c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>

take an isdn pcap dump of the wanpipe interface to see the messages there (I
think you already know how, if not, search "pcap" in wiki.sangoma.com)

On Mon, Mar 15, 2010 at 9:51 AM, Tzury Bar Yochay
<tzury.by at reguluslabs.com>wrote:

> Hi,
>
> This is my setup
>
> FreeSWITCH: Version 1.0.trunk (16992M)
> Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58
>
> I have finally managed to set up the environment coping many hick-ups
> reported in this mailing list recently.
> However, I am still encountering problems when trying to perform call
> from SIP to PRI.
>
> I have configured my dialplan to bridge all numbers which are not SIP
> to be bridged over to the PSTN.
> However, when calling a PSTN number I am getting
> RECOVERY_ON_TIMER_EXPIRE error
>
> pastebin for all configuration files is available at:
> http://pastebin.freeswitch.org/12437
>
> pastebin for CLI output is available at:
> http://pastebin.freeswitch.org/12436
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From lukasz at voiceworks.pl  Mon Mar 15 07:11:51 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 15 Mar 2010 15:11:51 +0100
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
 using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
Message-ID: <4B9E4027.4000100@voiceworks.pl>

I would check wanpipe 3.5.10.9 first. I think I had similar problem when 
I installed new openzap and older sangoma_prid (older wanpipe).

Tzury Bar Yochay wrote:
> Hi,
>
> This is my setup
>
> FreeSWITCH: Version 1.0.trunk (16992M)
> Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58
>
> I have finally managed to set up the environment coping many hick-ups
> reported in this mailing list recently.
> However, I am still encountering problems when trying to perform call
> from SIP to PRI.
>
> I have configured my dialplan to bridge all numbers which are not SIP
> to be bridged over to the PSTN.
> However, when calling a PSTN number I am getting
> RECOVERY_ON_TIMER_EXPIRE error
>
> pastebin for all configuration files is available at:
> http://pastebin.freeswitch.org/12437
>
> pastebin for CLI output is available at:
> http://pastebin.freeswitch.org/12436
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From tzury.by at reguluslabs.com  Mon Mar 15 07:11:58 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 16:11:58 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>
Message-ID: <10128ef11003150711q571f28fescf478066e3b3b209@mail.gmail.com>

On Mon, Mar 15, 2010 at 4:03 PM, Moises Silva <moises.silva at gmail.com> wrote:
> take an isdn pcap dump of the wanpipe interface to see the messages there (I
> think you already know how, if not, search "pcap" in wiki.sangoma.com)
pcap file attached, also available at
http://rapidshare.com/files/363660904/isdn.pcap.html
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From tzury.by at reguluslabs.com  Mon Mar 15 07:21:44 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 16:21:44 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <4B9E4027.4000100@voiceworks.pl>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<4B9E4027.4000100@voiceworks.pl>
Message-ID: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>

On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl> wrote:
> I would check wanpipe 3.5.10.9 first. I think I had similar problem when
> I installed new openzap and older sangoma_prid (older wanpipe).

Bu then I might face this issue
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
which was the reason I went to this version at first place


From lloyd.aloysius at gmail.com  Mon Mar 15 07:34:22 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 10:34:22 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003150956.37356.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>

How to execute another lua script when channel enter the REPORTING state?

What are the channel variable names?

Thanks
Lloyd

On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Unfortunately all "interesting" billing-related channel variables are set
> after the lua script execution, when channel enters REPORTING state. To
> access
> those variables I execute separate lua script using mod_cdr_csv, it allows
> to
> invoke lua script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > Hi All,
> >
> > I have the following simple Lua script answer a inbound call then dial
> > another number and bridge the call.
> >
> > Is there any way to get the  call duration or billing sec using the
> >
> > session:getVariable(  );
> >
> >
> >
> > -- answer the call
> > session:answer();
> >
> > -- sleep a second
> > session:sleep(1000);
> >
> > -- Initiate an outbound call
> > obSession =
> freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> >
> > obSession:execute("sched_hangup","+60 alloted_timeout");
> >
> > -- bride the call
> >     freeswitch.bridge(session, obSession);
> >
> > -- hangup
> > session:hangup();
> >
> >
> > Thank you
> > Lloyd
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lukasz at voiceworks.pl  Mon Mar 15 07:36:47 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 15 Mar 2010 15:36:47 +0100
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
 using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
Message-ID: <4B9E45FF.3030204@voiceworks.pl>

I'm running

FreeSWITCH Version 1.0.trunk (16966)
WANPIPE Release: 3.5.10.9

in my lab and everything is ok. Check 
http://wiki.sangoma.com/wanpipe-freeswitch-install. They recommend 
wanpipe 3.5.10.9. I think you should try it.

Tzury Bar Yochay wrote:
> On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl> wrote:
>   
>> I would check wanpipe 3.5.10.9 first. I think I had similar problem when
>> I installed new openzap and older sangoma_prid (older wanpipe).
>>     
>
> Bu then I might face this issue
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
> which was the reason I went to this version at first place
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From sos at sokhapkin.dyndns.org  Mon Mar 15 07:48:34 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 10:48:34 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
Message-ID: <201003151048.34689.sos@sokhapkin.dyndns.org>

cdr_csv.conf.xml:

<param name="default-template" value="lua"/>
<templates>
  <template name="lua">${lua(onhangup.lua)}</template>

All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv are 
accessible from that lua script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> How to execute another lua script when channel enter the REPORTING state?
> 
> What are the channel variable names?
> 
> Thanks
> Lloyd
> 
> On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Unfortunately all "interesting" billing-related channel variables are set
> > after the lua script execution, when channel enters REPORTING state. To
> > access
> > those variables I execute separate lua script using mod_cdr_csv, it
> > allows to
> > invoke lua script.
> >
> > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > Hi All,
> > >
> > > I have the following simple Lua script answer a inbound call then dial
> > > another number and bridge the call.
> > >
> > > Is there any way to get the  call duration or billing sec using the
> > >
> > > session:getVariable(  );
> > >
> > >
> > >
> > > -- answer the call
> > > session:answer();
> > >
> > > -- sleep a second
> > > session:sleep(1000);
> > >
> > > -- Initiate an outbound call
> > > obSession =
> >
> > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> >
> > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > >
> > > -- bride the call
> > >     freeswitch.bridge(session, obSession);
> > >
> > > -- hangup
> > > session:hangup();
> > >
> > >
> > > Thank you
> > > Lloyd
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From ahmed.ajmal at breezecom.ae  Mon Mar 15 07:50:49 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Mon, 15 Mar 2010 19:50:49 +0500
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
References: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
	<D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
Message-ID: <016201cac44e$eadf33f0$c09d9bd0$@ajmal@breezecom.ae>

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Thanks

Ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, March 10, 2010 7:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

We already set you a variable called sip_gateway_name when a call is placed
out a gateway.  You're also doing this wrong you should NEVER EVER do
@$ipgwX as that data is in the gateway.  You should simply do
sofia/gateway/gwname/number

 

/b

 

On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:





Hi

 

I am trying to set a channel variable on the bridge application using this:

 

<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>

 

The application runs fine, what I now need to do is set/export the 'gw'
channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.

 

Thanks

Ahmed

 

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From lloyd.aloysius at gmail.com  Mon Mar 15 07:54:13 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 10:54:13 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003151048.34689.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>

How to trigger this template from the main script ? or Does this
automatically trigger when finish the first script complete?

Thanks
Lloyd

On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> cdr_csv.conf.xml:
>
> <param name="default-template" value="lua"/>
> <templates>
>  <template name="lua">${lua(onhangup.lua)}</template>
>
> All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv are
> accessible from that lua script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > How to execute another lua script when channel enter the REPORTING state?
> >
> > What are the channel variable names?
> >
> > Thanks
> > Lloyd
> >
> > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Unfortunately all "interesting" billing-related channel variables are
> set
> > > after the lua script execution, when channel enters REPORTING state. To
> > > access
> > > those variables I execute separate lua script using mod_cdr_csv, it
> > > allows to
> > > invoke lua script.
> > >
> > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > Hi All,
> > > >
> > > > I have the following simple Lua script answer a inbound call then
> dial
> > > > another number and bridge the call.
> > > >
> > > > Is there any way to get the  call duration or billing sec using the
> > > >
> > > > session:getVariable(  );
> > > >
> > > >
> > > >
> > > > -- answer the call
> > > > session:answer();
> > > >
> > > > -- sleep a second
> > > > session:sleep(1000);
> > > >
> > > > -- Initiate an outbound call
> > > > obSession =
> > >
> > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > >
> > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > >
> > > > -- bride the call
> > > >     freeswitch.bridge(session, obSession);
> > > >
> > > > -- hangup
> > > > session:hangup();
> > > >
> > > >
> > > > Thank you
> > > > Lloyd
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
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From sos at sokhapkin.dyndns.org  Mon Mar 15 08:02:03 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 11:02:03 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
Message-ID: <201003151102.03407.sos@sokhapkin.dyndns.org>

That script will be executed automatically when call ends. You can set some 
channel variables in the main lua script and access those variables from 
hangup lua script if you need to pass the information to the hangup script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> How to trigger this template from the main script ? or Does this
> automatically trigger when finish the first script complete?
> 
> Thanks
> Lloyd
> 
> On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > cdr_csv.conf.xml:
> >
> > <param name="default-template" value="lua"/>
> > <templates>
> >  <template name="lua">${lua(onhangup.lua)}</template>
> >
> > All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv
> > are accessible from that lua script.
> >
> > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > How to execute another lua script when channel enter the REPORTING
> > > state?
> > >
> > > What are the channel variable names?
> > >
> > > Thanks
> > > Lloyd
> > >
> > > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Unfortunately all "interesting" billing-related channel variables are
> >
> > set
> >
> > > > after the lua script execution, when channel enters REPORTING state.
> > > > To access
> > > > those variables I execute separate lua script using mod_cdr_csv, it
> > > > allows to
> > > > invoke lua script.
> > > >
> > > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > > Hi All,
> > > > >
> > > > > I have the following simple Lua script answer a inbound call then
> >
> > dial
> >
> > > > > another number and bridge the call.
> > > > >
> > > > > Is there any way to get the  call duration or billing sec using the
> > > > >
> > > > > session:getVariable(  );
> > > > >
> > > > >
> > > > >
> > > > > -- answer the call
> > > > > session:answer();
> > > > >
> > > > > -- sleep a second
> > > > > session:sleep(1000);
> > > > >
> > > > > -- Initiate an outbound call
> > > > > obSession =
> > > >
> > > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > > >
> > > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > > >
> > > > > -- bride the call
> > > > >     freeswitch.bridge(session, obSession);
> > > > >
> > > > > -- hangup
> > > > > session:hangup();
> > > > >
> > > > >
> > > > > Thank you
> > > > > Lloyd
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> 



From moises.silva at gmail.com  Mon Mar 15 08:28:13 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 11:28:13 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
Message-ID: <c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>

Ok, so the call is not even being placed in the ISDN network.

Did you manage to get the sangoma_prid logs working? if that's the case then
paste them too, it seems sangoma_prid is not seeing the request from
FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
response from sangoma_prid binary (they both communicate through sctp
socket).
-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Mon, Mar 15, 2010 at 10:21 AM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

> On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl>
> wrote:
> > I would check wanpipe 3.5.10.9 first. I think I had similar problem when
> > I installed new openzap and older sangoma_prid (older wanpipe).
>
> Bu then I might face this issue
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
> which was the reason I went to this version at first place
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Mar 15 08:31:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 09:31:03 -0600
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
Message-ID: <191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>

you could wait for the execute_complete event for your break command.


On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  I?m not sure if this belongs in Jira, or if it?s possible to do it
> another way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From srinivas.ksvreddy at gmail.com  Mon Mar 15 08:36:29 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 15 Mar 2010 11:36:29 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
Message-ID: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>

Hi,

i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
like just farwading the call.
<action application="bridge" data=
sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
/ <data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>>
i do not using any gateway here, just farwarding call to router, is this
correct way of doing? any idea

-- 
Srinivasula Reddy K
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From peter.olsson at visionutveckling.se  Mon Mar 15 08:40:48 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Mar 2010 16:40:48 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>

Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.

On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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



--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
!DSPAM:4b9e543132932880416179!
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From anthony.minessale at gmail.com  Mon Mar 15 08:45:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 09:45:37 -0600
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
Message-ID: <191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>

That seems suspiciously similar to an old issue we have already fixed, how
exactly did you rebuild?


On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I rebuilt today.
>
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>
> > Are you sure you're on SVN trunk?
> >
> > /b
> >
> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
> >
> >> No.  Its on our hardware, we just use xen to virtualize the
> >> machines.. this is the only vm on the machine.  Gcore complains
> >> about no debugging symbols found, do I need to rebuild?
> >>
> >> It appears to be a problem when registering to one of my trunks.
> >> The very strange thing is the everything will be working perfectly
> >> and then all of a sudden this happens, i.e. trunks regged, calls
> >> completing, etc. This is a very lightly loaded server.
> >>
> >> This is the output from the console:
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar 15 09:21:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:21:15 -0600
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
Message-ID: <191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>

you could also wait for the execute_complete of the file you were playing

On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
>  On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> !DSPAM:4b9e543132932880416179!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Mar 15 09:28:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:28:05 -0600
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
Message-ID: <191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>

you would have to execute

dl_debug on

to get a trace that would tell us anything about that?

Are you using the windows gtalk client or the web site?
We are following google who have been pushing for the online client embedded
in the gmail website.




On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:

> here is the client:
>
> <profile type="client">
>  <param name="name" value="gmail.com"/>
>  <param name="login" value="xx at gmail.com/talk"/>
>  <param name="password" value="password"/>
>  <param name="server" value="talk.google.com"/>
>  <param name="dialplan" value="XML"/>
>  <param name="contex" value="public"/>
>  <param name="message" value="FreeSWITCH"/>
>  <param name="rtp-ip" value="auto"/>
>  <!-- if you are behind NAT, you should probably use stun -->
>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>  <param name="auto-login" value="true"/>
>  <param name="auto-reply" value="Welcome"/>
>  <param name="sasl" value="plain"/>
>  <param name="tls" value="true"/>
>  <param name="exten" value="5000"/>
> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> </profile>
>
> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> > ALL,
> >
> > I just upgrade FS to 16992.
> >
> > 1) timer warning: How's possible the 10000 microseconds? audio sounds ok.
> >
> >
> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> detected!
> > Do you have your kernel timer set to higher than 1 kHz? You may
> > experience audio problems.
> >
> > uname -a
> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> > 2008 i686 GNU/Linux
> >
> >
> > I used a script to check the interupts
> >
> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> > LOC:   14892854   14892843   14892841   14892830
> > LOC:   14892955   14892944   14892942   14892931
> > LOC:   14893056   14893045   14893043   14893032
> > LOC:   14893157   14893146   14893144   14893133
> > LOC:   14893258   14893247   14893245   14893234
> > LOC:   14893359   14893348   14893346   14893335
> > LOC:   14893460   14893449   14893447   14893436
> >
> >
> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> > originate a call from FS, the console blocks, gtalk client ring and it
> > looks ok when I hit answer, but no sound, FS console still blocks. a
> > few seconds later gtalk client shows:
> >
> > Sorry! The voice chat with language_lesson failed because of a network
> > problem at 5:50 PM. Please try again.
> >  Click here to report this to Google.
> >
> >
> > here is a log with bgapi:
> > http://pastebin.freeswitch.org/12434
> >
> >
> > rev 14696 works on the same server same conf. another server running
> > 16958M works at the same time. I have no clue to check, is it related
> > to the timer as in 1)? can some help me to take a look? let me know if
> > you need "dl_debug on" logs.
> >
> > Thanks.
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Mon Mar 15 09:39:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:39:26 -0600
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268643396699-4735736.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
Message-ID: <191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>

according to the code:

you are bridging 2 channels where one of them is not answered yet.
and if it never is answered before the timeout it will hangup.

try waiting for both to be answered before you bridge them.



On Mon, Mar 15, 2010 at 2:56 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello,
> I have the following problem:
>
> I register with x-lite to my freeswitch and then call it by sending command
> bgapi originate
>
> {originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
> &park()
>
> Then i dial through a gateway which points to another freeswitch instance:
> bgapi originate
>
> {originate_timeout=20}[24cc_session_id=3134a5ae-ec00-4164-94b1-a7fb5e0368fa,ignore_early_media=true,max_forwards=999999,origination_uuid=a85fd47e-051f-42ea-987e-f4e090c7ae78]sofia/gateway/prospect_gateway/1300671139878747511456
> &park()
>
> The dialplan causes answering the phone, playing a file and hanging up.
>
> The two calls are connected with this command:
>
> bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
> intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline
>
>
> Unfortunately after the second call hangs up, the first call gets hanged up
> too.
>
> I recently upgraded to FS trunk.
>
> Logs are in pastebin:
> first freeswitch: http://pastebin.freeswitch.org/12432
> second freeswitch: http://pastebin.freeswitch.org/12433
>
> Please take a look and help me figure this out..
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4735736.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dujinfang at gmail.com  Mon Mar 15 10:01:08 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 01:01:08 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
Message-ID: <23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>

Thanks, It's late now, I will get logs tomorrow. I tested both on
Windows client and web site on Mac. They both work on the other two
instances of FreeSWITCH I mentioned in this thread.

btw, can you give me a hint on the timer warning?

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> you would have to execute
>
> dl_debug on
>
> to get a trace that would tell us anything about that?
>
> Are you using the windows gtalk client or the web site?
> We are following google who have been pushing for the online client embedded
> in the gmail website.
>
>
>
>
> On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> here is the client:
>>
>> <profile type="client">
>> ?<param name="name" value="gmail.com"/>
>> ?<param name="login" value="xx at gmail.com/talk"/>
>> ?<param name="password" value="password"/>
>> ?<param name="server" value="talk.google.com"/>
>> ?<param name="dialplan" value="XML"/>
>> ?<param name="contex" value="public"/>
>> ?<param name="message" value="FreeSWITCH"/>
>> ?<param name="rtp-ip" value="auto"/>
>> ?<!-- if you are behind NAT, you should probably use stun -->
>> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> ?<param name="auto-login" value="true"/>
>> ?<param name="auto-reply" value="Welcome"/>
>> ?<param name="sasl" value="plain"/>
>> ?<param name="tls" value="true"/>
>> ?<param name="exten" value="5000"/>
>> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> </profile>
>>
>> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> > ALL,
>> >
>> > I just upgrade FS to 16992.
>> >
>> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
>> > ok.
>> >
>> >
>> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> > detected!
>> > Do you have your kernel timer set to higher than 1 kHz? You may
>> > experience audio problems.
>> >
>> > uname -a
>> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
>> > 2008 i686 GNU/Linux
>> >
>> >
>> > I used a script to check the interupts
>> >
>> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >
>> >
>> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
>> > originate a call from FS, the console blocks, gtalk client ring and it
>> > looks ok when I hit answer, but no sound, FS console still blocks. a
>> > few seconds later gtalk client shows:
>> >
>> > Sorry! The voice chat with language_lesson failed because of a network
>> > problem at 5:50 PM. Please try again.
>> > ?Click here to report this to Google.
>> >
>> >
>> > here is a log with bgapi:
>> > http://pastebin.freeswitch.org/12434
>> >
>> >
>> > rev 14696 works on the same server same conf. another server running
>> > 16958M works at the same time. I have no clue to check, is it related
>> > to the timer as in 1)? can some help me to take a look? let me know if
>> > you need "dl_debug on" logs.
>> >
>> > Thanks.
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From maciej.aniserowicz at gmail.com  Mon Mar 15 10:04:04 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 15 Mar 2010 09:04:04 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
Message-ID: <1268672644810-4738333.post@n2.nabble.com>


Hi Anthony,
Sorry if this was unclear - I am actually waiting for the phone to be
answered. The two calls are bridged in handler for ANSWERED event of the 2nd
call...
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4738333.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Mar 15 10:04:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 09:04:46 -0800
Subject: [Freeswitch-users] not answering calls on TDM400 port
In-Reply-To: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>
References: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>
Message-ID: <87f2f3b91003151004j4c91f78cie800f9ce8be2eef4@mail.gmail.com>

The TDM400 is a zaptel/dahdi card so you can use zttool (or the dahdi
equivalent) to see what the channels are doing.
-MC

On Sat, Mar 13, 2010 at 5:53 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> I noticed Freeswitch wasn't answering calls on one of the TDM400 ports
> yesterday.
>
> All other ports seemed to be OK.
>
> Nothing showed up on the console, set at debug log level.
>
> I restarted Freeswitch and it came good.
>
> What should I do if it happens again? Is there any debugging
> information I can extract from the system? If so how?
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Mar 15 10:18:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 11:18:06 -0600
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
Message-ID: <191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>

FS would prefer you run your kernel at 1000hz you are probably on a VM if
its 10000hz


On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:

> Thanks, It's late now, I will get logs tomorrow. I tested both on
> Windows client and web site on Mac. They both work on the other two
> instances of FreeSWITCH I mentioned in this thread.
>
> btw, can you give me a hint on the timer warning?
>
> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > you would have to execute
> >
> > dl_debug on
> >
> > to get a trace that would tell us anything about that?
> >
> > Are you using the windows gtalk client or the web site?
> > We are following google who have been pushing for the online client
> embedded
> > in the gmail website.
> >
> >
> >
> >
> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> here is the client:
> >>
> >> <profile type="client">
> >>  <param name="name" value="gmail.com"/>
> >>  <param name="login" value="xx at gmail.com/talk"/>
> >>  <param name="password" value="password"/>
> >>  <param name="server" value="talk.google.com"/>
> >>  <param name="dialplan" value="XML"/>
> >>  <param name="contex" value="public"/>
> >>  <param name="message" value="FreeSWITCH"/>
> >>  <param name="rtp-ip" value="auto"/>
> >>  <!-- if you are behind NAT, you should probably use stun -->
> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >>  <param name="auto-login" value="true"/>
> >>  <param name="auto-reply" value="Welcome"/>
> >>  <param name="sasl" value="plain"/>
> >>  <param name="tls" value="true"/>
> >>  <param name="exten" value="5000"/>
> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> </profile>
> >>
> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> > ALL,
> >> >
> >> > I just upgrade FS to 16992.
> >> >
> >> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
> >> > ok.
> >> >
> >> >
> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> >> > detected!
> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> > experience audio problems.
> >> >
> >> > uname -a
> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> >> > 2008 i686 GNU/Linux
> >> >
> >> >
> >> > I used a script to check the interupts
> >> >
> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> > LOC:   14892854   14892843   14892841   14892830
> >> > LOC:   14892955   14892944   14892942   14892931
> >> > LOC:   14893056   14893045   14893043   14893032
> >> > LOC:   14893157   14893146   14893144   14893133
> >> > LOC:   14893258   14893247   14893245   14893234
> >> > LOC:   14893359   14893348   14893346   14893335
> >> > LOC:   14893460   14893449   14893447   14893436
> >> >
> >> >
> >> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> >> > originate a call from FS, the console blocks, gtalk client ring and it
> >> > looks ok when I hit answer, but no sound, FS console still blocks. a
> >> > few seconds later gtalk client shows:
> >> >
> >> > Sorry! The voice chat with language_lesson failed because of a network
> >> > problem at 5:50 PM. Please try again.
> >> >  Click here to report this to Google.
> >> >
> >> >
> >> > here is a log with bgapi:
> >> > http://pastebin.freeswitch.org/12434
> >> >
> >> >
> >> > rev 14696 works on the same server same conf. another server running
> >> > 16958M works at the same time. I have no clue to check, is it related
> >> > to the timer as in 1)? can some help me to take a look? let me know if
> >> > you need "dl_debug on" logs.
> >> >
> >> > Thanks.
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar 15 10:21:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 11:21:48 -0600
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268672644810-4738333.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
Message-ID: <191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>

As you can see in the log that call hangs up before you bridge it so it ends
in failure
you may want to wait and make sure the call makes it to park start instead
of answered before you bridge.


On Mon, Mar 15, 2010 at 11:04 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hi Anthony,
> Sorry if this was unclear - I am actually waiting for the phone to be
> answered. The two calls are bridged in handler for ANSWERED event of the
> 2nd
> call...
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4738333.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Mar 15 10:36:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 09:36:03 -0800
Subject: [Freeswitch-users] problem in sending calls to internal Gateway.
In-Reply-To: <249845.71206.qm@web65402.mail.ac4.yahoo.com>
References: <249845.71206.qm@web65402.mail.ac4.yahoo.com>
Message-ID: <87f2f3b91003151036s3d458580l91c4cec4d90d3e73@mail.gmail.com>

Prabin,

Thank you for posting the configs. It is difficult to say what is wrong at
this point because we don't know exactly what you are dialing. Could you
please use pastebin.freeswitch.org and give us some call examples? Capture
the debug output from the FreeSWITCH console. Also, be sure to say what
phone is making the call and what digits the phone is dialing. You might
actually be able to figure it out by looking at the debug output. If you are
using fs_cli then debug mode is on automatically. If you are at the actual
FreeSWITCH console (meaning that you did NOT start freeswitch with the -nc
flag) then be sure to press F8 or type "console loglevel debug" to enable
debug level output. See also this page for lots of handy tips on reporting
information for troubleshooting:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Sat, Mar 13, 2010 at 10:22 PM, Prabin Shrestha <nepaligas at yahoo.com>wrote:

> hello,
>
> My extension are working properly. But they can't send calls to gateway. I
> have 2 interfaces, one with public and another with private lan. Gateway is
> in private lan. Please suggest.
> My configurations are as follows:
>
> /conf/sip_profiles/external.xml:
>    <param name="rtp-ip" value="110.4.12.111"/>
>    <param name="sip-ip" value="110.4.12.111"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
>
> /conf/sip_profiles/internal.xml:
>     <param name="rtp-ip" value="192.168.1.11"/>
>     <param name="sip-ip" value="192.168.1.11"/>
>    <param name="hold-music" value="$${hold_music}"/>
>    <param name="apply-nat-acl" value="nat.auto"/>
>
> haven't made any changes in conf/autoload_configs/acl.conf.xml
>
> to route calls from providers, /dialplan/public/providers.xml:
>    <extension name="provider">
>    <condition field="network_addr" expression="200.144.44.177"/>
>    <condition field="destination_number" expression="^(\d+)$">
>        <action application="transfer" data="$1 XML context_AX"/>
>    </condition>
>   </extension>
>
> gateway is defined at /conf/sip_profiles/internal/gateways.xml:
>   <include>
>   <gateway name="gateway_AX">
>    <param name="username" value="admin"/>
>    <param name="password" value="nepal123"/>
>    <param name="proxy" value="192.168.1.100"/>
>    <param name="register" value="false"/>
>    <param name="extension" value="1001"/>
>    </gateway>
>   </include>
>
>
> To route internal calls from extensions, /conf/dialplan/default.xml:
> <extension name="my_internal">
>    <!--<condition field="${ruleset}" expression="internal" /> -->
>    <condition field="destination_number" expression="^(d+)$">
>      <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
>    </condition>
>  </extension>
>
> to route calls from providers, conf/dialplan/inbound_routing.xml:
> <context name="context_AX">
>    <extension name="from_provider_to_AX">
>    <!--<condition field="destination_number" expression="^00(1.*)$">-->
>    <condition field="destination_number" expression="^9(\d+)$">
>    <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
>    </condition>
>    </extension>
> </context>
>
> Please help.
>
>
>      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage.
> http://in.yahoo.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From robert.dyck at shaw.ca  Sun Mar 14 18:11:57 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Sun, 14 Mar 2010 18:11:57 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
Message-ID: <201003141811.58645.robert.dyck@shaw.ca>

16987 on Fedora

I realize fscomm is an immature project. I get an error message when it tries 
to detect my devices ( pci sound card, usb handset, usb camera with microphone 
). Is this the expected result? Detection works in Twinkle and Ekiga.

Second question - is video support on the roadmap?


From schogge at gmx.de  Mon Mar 15 08:14:02 2010
From: schogge at gmx.de (schogge)
Date: Mon, 15 Mar 2010 08:14:02 -0700 (PDT)
Subject: [Freeswitch-users]  FreeSWITCH to add SRTP to Asterisk
Message-ID: <27901964.post@talk.nabble.com>


Hi Community,
I'm new to FreeSWITCH. What I have read so far it seems to be a very
interesting project.
For our company we are running an Asterisk PBX in a datacenter. After some
years it is doing everything we want except of SRTP.
Would it be possible to run FreeSWITCH in front of Asterisk to convert SRTP
coming from our phones to RTP which is ok for Asterisk and vice versa?
Thanks,
schogge
-- 
View this message in context: http://old.nabble.com/FreeSWITCH-to-add-SRTP-to-Asterisk-tp27901964p27901964.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From infos at madovsky.org  Mon Mar 15 09:28:29 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 12:28:29 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>

Hi all,

is there any way to set the voicemail for leave a message without any option at the end ?
like record or send the message by email once the caller hangs up or after N sec of silence.

Thanks

Franck
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From anthony.minessale at gmail.com  Mon Mar 15 10:54:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 12:54:57 -0500
Subject: [Freeswitch-users] FreeSWITCH to add SRTP to Asterisk
In-Reply-To: <27901964.post@talk.nabble.com>
References: <27901964.post@talk.nabble.com>
Message-ID: <191c3a031003151054s28d84c38oc02ae3ddad38acf7@mail.gmail.com>

yes, it should be possible.


On Mon, Mar 15, 2010 at 10:14 AM, schogge <schogge at gmx.de> wrote:

>
> Hi Community,
> I'm new to FreeSWITCH. What I have read so far it seems to be a very
> interesting project.
> For our company we are running an Asterisk PBX in a datacenter. After some
> years it is doing everything we want except of SRTP.
> Would it be possible to run FreeSWITCH in front of Asterisk to convert SRTP
> coming from our phones to RTP which is ok for Asterisk and vice versa?
> Thanks,
> schogge
> --
> View this message in context:
> http://old.nabble.com/FreeSWITCH-to-add-SRTP-to-Asterisk-tp27901964p27901964.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From spencer at 5ninesolutions.com  Mon Mar 15 11:07:24 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 15 Mar 2010 11:07:24 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
	<191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
Message-ID: <5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>

I'm using Centos 5.4, I have a build machine and I got the latest  
source from SVN, and rebuilt the RPMS from the updates source and then  
updated the FS machine.  We have our own repo setup to update the  
machines.  I disabled most of the modules except mod_console,  
mod_logfile, mod_cdr_csv, mod_event_scoket, mod_sofia, mod_loopback,  
mod_commands, mod_dptools, mod_expr, mod_limit, mod_dialplan_xml, and  
mod_voipcodecs and everything is working great.  As soon as I enable  
the default set of modules, FS will work for about 3-4 hours and then  
one of the threads goes to 100% CPU.  I haven't had the time to load  
the modules one by one to try to isolate the problem.  The build  
completes fine, and I'm using the stock centos libraries to build  
against.  The centos xen kernel on x86_64 is 100hz, could that be an  
issue?


On Mar 15, 2010, at 8:45 AM, Anthony Minessale wrote:

> That seems suspiciously similar to an old issue we have already  
> fixed, how exactly did you rebuild?
>
>
> On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> I rebuilt today.
>
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>
> > Are you sure you're on SVN trunk?
> >
> > /b
> >
> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
> >
> >> No.  Its on our hardware, we just use xen to virtualize the
> >> machines.. this is the only vm on the machine.  Gcore complains
> >> about no debugging symbols found, do I need to rebuild?
> >>
> >> It appears to be a problem when registering to one of my trunks.
> >> The very strange thing is the everything will be working perfectly
> >> and then all of a sudden this happens, i.e. trunks regged, calls
> >> completing, etc. This is a very lightly loaded server.
> >>
> >> This is the output from the console:
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Mon Mar 15 11:18:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:18:39 -0800
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
	<3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
	<ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>
Message-ID: <87f2f3b91003151118u1a0373aete4903979e48aaab8@mail.gmail.com>

FYI,
I added this as example 16 on the dialplan xml page on the wiki.
-MC


2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> Wiki is your friend to that type of info! Contribute to it, please!
>
> JM
>
>
>
> On Sun, Mar 14, 2010 at 11:20 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> So just in case anyone else ever needs to do this.. here's what I ended up
>> with:
>>
>> <include>
>>   <extension name="NNNNNNNNNNN-did">
>>     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="set" data="accountcode=<ACCTNUM>"/>
>>       <action application="bridge"
>> data="sofia/default/$1${regex(${sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}"
>> />
>>     </condition>
>>   </extension>
>> </include>
>>
>> Basically the extension UAs register to is created as the same as the
>> account code.  We then use their ip at registration to dial a did on their
>> end.
>>
>> Spencer
>>
>>
>> On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:
>>
>> Duh.. :-)  Thanks!
>>
>> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>>
>> Yes, sofia_contact with regex to extract the IP portion? You had the
>> answer yourself.
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
>> spencer at 5ninesolutions.com> wrote:
>>
>>> I guess my question is, is there some function which will return the
>>> IP address of a registered user?  Similar to sofia_contact but only
>>> the IP address?
>>>
>>> Spencer
>>>
>>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>>
>>> > You sure can still have it call a registration.. You're mixing the
>>> > dialplan and user directly as if they are a single entity.  They
>>> > aren't.
>>> >
>>> > /b
>>> >
>>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>>> >
>>> >> That looks promising, thanks.  I was hoping to have FS do it
>>> >> automatically.  I.e. inbound dids are assigned to specific asterisk
>>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>>> >> inbound did @ whereever the registration came from instead of
>>> >> having the IP address hard coded into a route.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Mar 15 11:22:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 13:22:59 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
	<191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
	<5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>
Message-ID: <191c3a031003151122q601b8341s1f77172090ebc02f@mail.gmail.com>

if you can email me privately at consulting at freeswitch.org with the details
to log into the box.
I can have a look for you.


On Mon, Mar 15, 2010 at 1:07 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I'm using Centos 5.4, I have a build machine and I got the latest source
> from SVN, and rebuilt the RPMS from the updates source and then updated the
> FS machine.  We have our own repo setup to update the machines.  I disabled
> most of the modules except mod_console, mod_logfile, mod_cdr_csv,
> mod_event_scoket, mod_sofia, mod_loopback, mod_commands, mod_dptools,
> mod_expr, mod_limit, mod_dialplan_xml, and mod_voipcodecs and everything is
> working great.  As soon as I enable the default set of modules, FS will work
> for about 3-4 hours and then one of the threads goes to 100% CPU.  I haven't
> had the time to load the modules one by one to try to isolate the problem.
>  The build completes fine, and I'm using the stock centos libraries to build
> against.  The centos xen kernel on x86_64 is 100hz, could that be an issue?
>
>
> On Mar 15, 2010, at 8:45 AM, Anthony Minessale wrote:
>
> That seems suspiciously similar to an old issue we have already fixed, how
> exactly did you rebuild?
>
>
> On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I rebuilt today.
>>
>> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>>
>> > Are you sure you're on SVN trunk?
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
>> >
>> >> No.  Its on our hardware, we just use xen to virtualize the
>> >> machines.. this is the only vm on the machine.  Gcore complains
>> >> about no debugging symbols found, do I need to rebuild?
>> >>
>> >> It appears to be a problem when registering to one of my trunks.
>> >> The very strange thing is the everything will be working perfectly
>> >> and then all of a sudden this happens, i.e. trunks regged, calls
>> >> completing, etc. This is a very lightly loaded server.
>> >>
>> >> This is the output from the console:
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From msc at freeswitch.org  Mon Mar 15 11:23:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:23:02 -0800
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
Message-ID: <87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>

On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi all,
>
> is there any way to set the voicemail for leave a message without any
> option at the end ?
> like record or send the message by email once the caller hangs up or after
> N sec of silence.
>
>
I'm not quite sure that I understand the question. If the caller hangs up
while leaving a message then the message gets sent.
-MC
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From msc at freeswitch.org  Mon Mar 15 11:28:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:28:35 -0800
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <8259813661645760166@unknownmsgid>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
	<8259813661645760166@unknownmsgid>
Message-ID: <87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>wrote:

>  Hi
>
>
>
> I have now configured the gateways as suggested below but I am not able to
> see the sip_gateway_name variable when the call hangup event is complete. I
> am on Freeswitch 1.0.5.
>
>
>
Where are you looking to find the variable?
-MC
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From msc at freeswitch.org  Mon Mar 15 11:39:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:39:01 -0800
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <87f2f3b91003151139w5525dfa9pcb1d201bdb0ee91d@mail.gmail.com>

On Mon, Mar 15, 2010 at 7:36 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
> like just farwading the call.
>

1.0.2? Are you running it on MS-DOS 2.0?!
Seriously, you need to update. 1.0.2 had some bugs. We're now at 1.0.5
(essentially). 1.0.2 is like 3 generations old.

> <action application="bridge" data=<data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>
> sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
> />
> i do not using any gateway here, just farwarding call to router, is this
> correct way of doing? any idea
>

You might want to use $1 instead of ${destination_number} since $1 will
always have the right information in it, assuming of course that you used (
and ) in your regular expression. Something like this ought to work

<action application="bridge" data="sofia/internal/$1 at 1.2.3.4"/> <!-- 1.2.3.4
is the gateway IP addr -->

-MC
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From lloyd.aloysius at gmail.com  Mon Mar 15 11:50:25 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 14:50:25 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003151102.03407.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>

How to control the default-template. I do not want to run all the time when
call hangup. I want only when I need for some hangup information..

Thanks
Lloyd


On Mon, Mar 15, 2010 at 11:02 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> That script will be executed automatically when call ends. You can set some
> channel variables in the main lua script and access those variables from
> hangup lua script if you need to pass the information to the hangup script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > How to trigger this template from the main script ? or Does this
> > automatically trigger when finish the first script complete?
> >
> > Thanks
> > Lloyd
> >
> > On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > cdr_csv.conf.xml:
> > >
> > > <param name="default-template" value="lua"/>
> > > <templates>
> > >  <template name="lua">${lua(onhangup.lua)}</template>
> > >
> > > All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv
> > > are accessible from that lua script.
> > >
> > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > How to execute another lua script when channel enter the REPORTING
> > > > state?
> > > >
> > > > What are the channel variable names?
> > > >
> > > > Thanks
> > > > Lloyd
> > > >
> > > > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> > > >
> > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > Unfortunately all "interesting" billing-related channel variables
> are
> > >
> > > set
> > >
> > > > > after the lua script execution, when channel enters REPORTING
> state.
> > > > > To access
> > > > > those variables I execute separate lua script using mod_cdr_csv, it
> > > > > allows to
> > > > > invoke lua script.
> > > > >
> > > > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > > > Hi All,
> > > > > >
> > > > > > I have the following simple Lua script answer a inbound call then
> > >
> > > dial
> > >
> > > > > > another number and bridge the call.
> > > > > >
> > > > > > Is there any way to get the  call duration or billing sec using
> the
> > > > > >
> > > > > > session:getVariable(  );
> > > > > >
> > > > > >
> > > > > >
> > > > > > -- answer the call
> > > > > > session:answer();
> > > > > >
> > > > > > -- sleep a second
> > > > > > session:sleep(1000);
> > > > > >
> > > > > > -- Initiate an outbound call
> > > > > > obSession =
> > > > >
> > > > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > > > >
> > > > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > > > >
> > > > > > -- bride the call
> > > > > >     freeswitch.bridge(session, obSession);
> > > > > >
> > > > > > -- hangup
> > > > > > session:hangup();
> > > > > >
> > > > > >
> > > > > > Thank you
> > > > > > Lloyd
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> >
>
>
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From msc at freeswitch.org  Mon Mar 15 11:52:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:52:31 -0800
Subject: [Freeswitch-users] FreeSWITCH Conf Call Agenda
Message-ID: <87f2f3b91003151152s1acc22f4ydf45f2e6630673b3@mail.gmail.com>

Hello all,

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

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

Thanks!
-MC
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From infos at madovsky.org  Mon Mar 15 12:18:10 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 15:18:10 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
Message-ID: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>

Actually on the default voicemail example 
after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
so if the caller hangs up before to press 2 the message is not sent. any way to remove
these options and send the message after slience or caller hang up ?

Thanks

F
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 2:23 PM
  Subject: Re: [Freeswitch-users] voicemail options





  On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

    Hi all,

    is there any way to set the voicemail for leave a message without any option at the end ?
    like record or send the message by email once the caller hangs up or after N sec of silence.

  I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
  -MC





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From testeador01 at gmail.com  Mon Mar 15 12:32:04 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 15 Mar 2010 14:32:04 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
Message-ID: <c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>

Hello,

I have voicemails set on my office and messages do get sent when you hang up
even if you don't press 2,

did you try it? or are you guessing? (not trying to be rude, just trying to
get info :) )

what revision are you running?



2010/3/15 Madovsky <infos at madovsky.org>

>  Actually on the default voicemail example
> after you left a message the operator says press 1 to listen blabla or 2 to
> save blabla..
> so if the caller hangs up before to press 2 the message is not sent. any
> way to remove
> these options and send the message after slience or caller hang up ?
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 15, 2010 2:23 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
>
>
> On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  Hi all,
>>
>> is there any way to set the voicemail for leave a message without any
>> option at the end ?
>> like record or send the message by email once the caller hangs up or after
>> N sec of silence.
>>
>>
> I'm not quite sure that I understand the question. If the caller hangs up
> while leaving a message then the message gets sent.
> -MC
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Mar 15 13:01:38 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:01:38 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
Message-ID: <C3263B8079914E40A8653B84DA785598@MOBILEE1705>

Hi,

ah ok, so maybe the reason of fail is this error in console

2010-03-15 16:01:36.752887 [CRIT] switch_channel.c:759 Invalid data (${RECORD_COMMENT} contains a variable)

Any idea ?

Thanks
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 3:32 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


  did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


  what revision are you running?






  2010/3/15 Madovsky <infos at madovsky.org>

    Actually on the default voicemail example 
    after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
    so if the caller hangs up before to press 2 the message is not sent. any way to remove
    these options and send the message after slience or caller hang up ?

    Thanks

    F
      ----- Original Message ----- 
      From: Michael Collins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 2:23 PM
      Subject: Re: [Freeswitch-users] voicemail options





      On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

        Hi all,

        is there any way to set the voicemail for leave a message without any option at the end ?
        like record or send the message by email once the caller hangs up or after N sec of silence.

      I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
      -MC





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


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



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






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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 13:02:05 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:02:05 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
Message-ID: <A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>

Sorry revision 16970
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 3:32 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


  did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


  what revision are you running?






  2010/3/15 Madovsky <infos at madovsky.org>

    Actually on the default voicemail example 
    after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
    so if the caller hangs up before to press 2 the message is not sent. any way to remove
    these options and send the message after slience or caller hang up ?

    Thanks

    F
      ----- Original Message ----- 
      From: Michael Collins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 2:23 PM
      Subject: Re: [Freeswitch-users] voicemail options





      On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

        Hi all,

        is there any way to set the voicemail for leave a message without any option at the end ?
        like record or send the message by email once the caller hangs up or after N sec of silence.

      I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
      -MC





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


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



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






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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From testeador01 at gmail.com  Mon Mar 15 13:10:54 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 15 Mar 2010 15:10:54 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
	<A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
Message-ID: <c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>

Hello,

What value are you setting for "RECORD_COMMENT"? what is your dial plan for
the extension you're testing?



2010/3/15 Madovsky <infos at madovsky.org>

>  Sorry revision 16970
>
> ----- Original Message -----
> *From:* Milena <testeador01 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 15, 2010 3:32 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
> Hello,
>
> I have voicemails set on my office and messages do get sent when you hang
> up even if you don't press 2,
>
> did you try it? or are you guessing? (not trying to be rude, just trying to
> get info :) )
>
> what revision are you running?
>
>
>
> 2010/3/15 Madovsky <infos at madovsky.org>
>
>>  Actually on the default voicemail example
>> after you left a message the operator says press 1 to listen blabla or 2
>> to save blabla..
>> so if the caller hangs up before to press 2 the message is not sent. any
>> way to remove
>> these options and send the message after slience or caller hang up ?
>>
>> Thanks
>>
>> F
>>
>>   ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Monday, March 15, 2010 2:23 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>>
>>
>> On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  Hi all,
>>>
>>> is there any way to set the voicemail for leave a message without any
>>> option at the end ?
>>> like record or send the message by email once the caller hangs up or
>>> after N sec of silence.
>>>
>>>
>> I'm not quite sure that I understand the question. If the caller hangs up
>> while leaving a message then the message gets sent.
>> -MC
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
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>  ------------------------------
>
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From xengelpublicx at gmail.com  Mon Mar 15 13:21:07 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 15 Mar 2010 23:21:07 +0300
Subject: [Freeswitch-users] intercept in fifo queue
Message-ID: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>

Hello.

How can I intercept a fifo? I tried to use the example of intecept
default dialplan, but in no fifo dialed_extension.

    <extension name="operators">
      <condition field="destination_number" expression="^operators$">
            <action application="bind_meta_app" data="2 b s
record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
            <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
            <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
            <action application="answer"/>
            <action application="playback"
data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
            <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>
            <action application="set" data="fifo_music=$${hold_music}"/>
            <action application="fifo" data="operators@$${domain} in undef"/>
      </condition>
    </extension>

    <fifo name="operators@$${domain}" importance="0">
      <member timeout="15" simo="1"
lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member>
    </fifo>
  </fifos>
-- 
Best regards, Vladimir Elizarov


From robert.hadley at teotech.com  Mon Mar 15 13:24:07 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 15 Mar 2010 13:24:07 -0700
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
Message-ID: <6CFB240D4D064C63BB295CAADAD1EF9E@greyhawk.tonecommander.com>

 

The default FS action is to save the voicemail, so it is saved when there is
silence or the caller hangs up without pressing 2.  Note there is a minimum
recording length of several seconds before a VM is saved.

 

Robert

 

  _____  

From: Madovsky [mailto:infos at madovsky.org] 
Sent: Monday, March 15, 2010 12:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] voicemail options

 

Actually on the default voicemail example 

after you left a message the operator says press 1 to listen blabla or 2 to
save blabla..

so if the caller hangs up before to press 2 the message is not sent. any way
to remove

these options and send the message after slience or caller hang up ?

 

Thanks

 

F

----- Original Message ----- 

From: Michael Collins <mailto:msc at freeswitch.org>  

To: freeswitch-users at lists.freeswitch.org 

Sent: Monday, March 15, 2010 2:23 PM

Subject: Re: [Freeswitch-users] voicemail options

 

 

On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

Hi all,

 

is there any way to set the voicemail for leave a message without any option
at the end ?

like record or send the message by email once the caller hangs up or after N
sec of silence.

 

I'm not quite sure that I understand the question. If the caller hangs up
while leaving a message then the message gets sent. 
-MC

 


  _____  


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 13:24:56 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:24:56 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
Message-ID: <5B1C9610DC644E8390EA3B9D450F4DBB@MOBILEE1705>

 here is my dialplan

<extension name="phone_connection" continue="true">
   <condition field="destination_number" expression="^(123456789)$">
   <action application="set" data="dialed_extension=$1"/>
   <action application="export" data="dialed_extension=$1"/>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="call_timeout=15"/>
   <action application="bridge" data="{monitor_early_media_fail=user_busy:2:480+620!user_not_registered:2:1776.7!subscriber_absent:2:1776.7,!no_user_response:2:1776.7,sip_from_uri=${sip_from_user}@${domain}}user/${dialed_extension}@${domain}"/>
   <action application="set" data="call_timeout=15"/>
   <action application="set" data="originate_timeout=5"/>
   <action application="bridge" data="{monitor_early_media_fail=user_busy:2:480+620!user_not_registered:2:1776.7!subscriber_absent:2:1776.7,!no_user_response:2:1776.7,origination_caller_id_name=${outbound_caller_id_name},origination_caller_id_number=${outbound_caller_id}}sofia/gateway/${distributor(pstn_domestic)}/00${destination_number}|sofia/gateway/${distributor(pstn_international_2)}/011${destination_number}" loop="2"/>
   <action application="answer"/>
   <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain} ${dialed_extension}"/>

absolutly have no idea where RECORD_COMMENT var comes from...

Thx
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  What value are you setting for "RECORD_COMMENT"? what is your dial plan for the extension you're testing?






  2010/3/15 Madovsky <infos at madovsky.org>

    Sorry revision 16970
      ----- Original Message ----- 
      From: Milena 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 3:32 PM
      Subject: Re: [Freeswitch-users] voicemail options


      Hello,


      I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


      did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


      what revision are you running?






      2010/3/15 Madovsky <infos at madovsky.org>

        Actually on the default voicemail example 
        after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
        so if the caller hangs up before to press 2 the message is not sent. any way to remove
        these options and send the message after slience or caller hang up ?

        Thanks

        F
          ----- Original Message ----- 
          From: Michael Collins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 2:23 PM
          Subject: Re: [Freeswitch-users] voicemail options





          On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

            Hi all,

            is there any way to set the voicemail for leave a message without any option at the end ?
            like record or send the message by email once the caller hangs up or after N sec of silence.

          I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
          -MC





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


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



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


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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 13:38:42 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:38:42 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<6CFB240D4D064C63BB295CAADAD1EF9E@greyhawk.tonecommander.com>
Message-ID: <AE28A09C45E94D99804FBD5B5256E738@MOBILEE1705>

yes by

      <param name="min-record-len" value="5"/>
      <param name="max-record-len" value="180"/>

was not sure that this save behavior was in default FS action.

Thanks

F
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:24 PM
  Subject: Re: [Freeswitch-users] voicemail options


   

  The default FS action is to save the voicemail, so it is saved when there is silence or the caller hangs up without pressing 2.  Note there is a minimum recording length of several seconds before a VM is saved.

   

  Robert

   


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

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Monday, March 15, 2010 12:18 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] voicemail options

   

  Actually on the default voicemail example 

  after you left a message the operator says press 1 to listen blabla or 2 to save blabla..

  so if the caller hangs up before to press 2 the message is not sent. any way to remove

  these options and send the message after slience or caller hang up ?

   

  Thanks

   

  F

    ----- Original Message ----- 

    From: Michael Collins 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Monday, March 15, 2010 2:23 PM

    Subject: Re: [Freeswitch-users] voicemail options

     

     

    On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

    Hi all,

     

    is there any way to set the voicemail for leave a message without any option at the end ?

    like record or send the message by email once the caller hangs up or after N sec of silence.

     

    I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
    -MC

     


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

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From infos at madovsky.org  Mon Mar 15 13:59:37 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:59:37 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
Message-ID: <180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>

ok maybe this is the var from voicemail.conf.xml record-comment ?
      <param name="record-comment" value="message from a call at ${domain}"/>
didn't know it doesn't allow variables... so I removed it and the error disappeared.
so I have on console

2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz

but no message sent by email yet..
I checked sendmail -t manually, it works.

here is my voicemail.conf.xml

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="mp3"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="5"/>
      <param name="max-record-len" value="180"/>
      <param name="max-retries" value="3"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="login-keys" value="0"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="skip-greet-key" value="#"/>
      <!--<param name="previous-message-key" value=""/>-->
      <!--<param name="next-message-key" value=""/>-->
      <!--<param name="skip-info-key" value=""/>-->
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <param name="storage-dir" value="/tmp"/>
      <param name="odbc-dsn" value="freeswitch-pgsql:freeswitch:bronto8fc"/>
      <param name="record-comment" value="message from a call"/>
      <param name="record-title" value="your Boophone voicemail"/>
      <param name="record-copyright" value="own Copyright"/>
    </profile>
  </profiles>
</configuration>


and the user config
....
    <params>
      ....
     <param name="vm-email-all-messages" value="true"/>
     <param name="vm-attach-file" value="true"/>
     <param name="vm-keep-local-after-email" value="false"/>
      <param name="vm-mailto" value="infos at baba.bibi"/>
...
    </params>


F



  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  What value are you setting for "RECORD_COMMENT"? what is your dial plan for the extension you're testing?






  2010/3/15 Madovsky <infos at madovsky.org>

    Sorry revision 16970
      ----- Original Message ----- 
      From: Milena 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 3:32 PM
      Subject: Re: [Freeswitch-users] voicemail options


      Hello,


      I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


      did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


      what revision are you running?






      2010/3/15 Madovsky <infos at madovsky.org>

        Actually on the default voicemail example 
        after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
        so if the caller hangs up before to press 2 the message is not sent. any way to remove
        these options and send the message after slience or caller hang up ?

        Thanks

        F
          ----- Original Message ----- 
          From: Michael Collins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 2:23 PM
          Subject: Re: [Freeswitch-users] voicemail options





          On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

            Hi all,

            is there any way to set the voicemail for leave a message without any option at the end ?
            like record or send the message by email once the caller hangs up or after N sec of silence.

          I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
          -MC





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


          _______________________________________________
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          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 14:14:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 16:14:52 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
	<180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
Message-ID: <2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>

Word of advice.  MP3 is not recommended at all.

/b

On Mar 15, 2010, at 3:59 PM, Madovsky wrote:

> 2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
> 2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz

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From msc at freeswitch.org  Mon Mar 15 14:15:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 13:15:03 -0800
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
Message-ID: <87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>

On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> How to control the default-template. I do not want to run all the time when
> call hangup. I want only when I need for some hangup information..
>
> Thanks
> Lloyd
>

I don't know that you can selectively run or not run a script at hangup
time. However, since Lua is so lightweight it probably would be fine just to
run it at each hangup.
-MC
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From lloyd.aloysius at gmail.com  Mon Mar 15 14:38:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 17:38:24 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
	<87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
Message-ID: <8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>

I would like to do some Database insert and update operations , Not for all
the calls only the calls originating from a particular script.







On Mon, Mar 15, 2010 at 5:15 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> How to control the default-template. I do not want to run all the time
>> when call hangup. I want only when I need for some hangup information..
>>
>> Thanks
>> Lloyd
>>
>
> I don't know that you can selectively run or not run a script at hangup
> time. However, since Lua is so lightweight it probably would be fine just to
> run it at each hangup.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nazim.agabekov at gmail.com  Mon Mar 15 15:09:12 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 16 Mar 2010 02:09:12 +0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>	<201003151048.34689.sos@sokhapkin.dyndns.org>	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>	<201003151102.03407.sos@sokhapkin.dyndns.org>	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>	<87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
	<8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>
Message-ID: <4B9EB008.4080005@gmail.com>

Lloyd, if you need to run different Lua scripts depending on some
variables (or entirely skip the cdr), try my implementation.
Documentation is in very embryonic state. You could get the overview
from blog.buta-tech.com. Warning! all code snapshots on that page are
obsolete! Latetest svn trunk is at
http://nazim at svn.freeswitch.org/svn/freeswitch/trunk/contrib/nazim

Regards,
Nazim


On 03/16/2010 01:38 AM, Aloysius Lloyd wrote:
> I would like to do some Database insert and update operations , Not
> for all the calls only the calls originating from a particular script.
>
>
>
>
>
>
>
> On Mon, Mar 15, 2010 at 5:15 PM, Michael Collins <msc at freeswitch.org
> <mailto:msc at freeswitch.org>> wrote:
>
>
>
>     On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd
>     <lloyd.aloysius at gmail.com <mailto:lloyd.aloysius at gmail.com>> wrote:
>
>         How to control the default-template. I do not want to run all
>         the time when call hangup. I want only when I need for some
>         hangup information..
>
>         Thanks
>         Lloyd
>
>
>     I don't know that you can selectively run or not run a script at
>     hangup time. However, since Lua is so lightweight it probably
>     would be fine just to run it at each hangup.
>     -MC
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From will.traenkle at yahoo.com  Mon Mar 15 16:22:25 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 15 Mar 2010 16:22:25 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination fails
	// direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
Message-ID: <923176.22123.qm@web57601.mail.re1.yahoo.com>

When a hunt group is accessed from the main IVR, the hunt group dials
an internal extension via correctly but when trying to dial an
external number such a cell phone via sip uri it is unable to connect
or ring the cell phone and it waits the correct timeout period before it
goes to voicemail.

However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  


There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
 
5) Dialplan default.xml:  
http://pastebin.com/iAbNSW9E 
 
If you require additional information or have any other
questions, please let me know.

Thanks,

-Will

________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:00:53 PM
Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds




On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>
>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>
>Any ideas where I should look and what line of code I might be missing?
>
>

Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
-MC


      
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From lloydie.t at googlemail.com  Mon Mar 15 16:51:48 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Mon, 15 Mar 2010 23:51:48 +0000
Subject: [Freeswitch-users] Problem post request.
Message-ID: <8b61bd671003151651sda3970er7e2d2fcf50067ae2@mail.gmail.com>

I post two request re mod_xml_curl, but they both seemed to have
disappeared. Please can you advise how I can post request so that they can
be viewed as I am stuck on a few things?
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From infos at madovsky.org  Mon Mar 15 17:21:41 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 20:21:41 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
Message-ID: <82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>

Ok, for what reason ?
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 5:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Word of advice.  MP3 is not recommended at all.


  /b


  On Mar 15, 2010, at 3:59 PM, Madovsky wrote:


    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From brian at freeswitch.org  Mon Mar 15 17:31:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 19:31:14 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
	<82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
Message-ID: <306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>

You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.

/b

On Mar 15, 2010, at 7:21 PM, Madovsky wrote:

> Ok, for what reason ?
> 

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From infos at madovsky.org  Mon Mar 15 17:46:27 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 20:46:27 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
Message-ID: <EA0AE9A51FD940A7B920337A3DCCE5F1@MOBILEE1705>

ok I understand.
So any way to avoid wav (which could be big for an email) ?
I guess I can use the audio format of modules I compiled in FS isn't it ?

  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 8:31 PM
  Subject: Re: [Freeswitch-users] voicemail options


  You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.


  /b


  On Mar 15, 2010, at 7:21 PM, Madovsky wrote:


    Ok, for what reason ?






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From spencer at 5ninesolutions.com  Mon Mar 15 17:55:56 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 15 Mar 2010 17:55:56 -0700
Subject: [Freeswitch-users] Radius CDR
Message-ID: <BE35BA86-E45D-4BCD-8AB7-4BD40BE561E5@5ninesolutions.com>

Hello all,
Is mod_radius_cdr still around and does anyone have any experience  
with it?

Spencer



From infos at madovsky.org  Mon Mar 15 19:26:04 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 22:26:04 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
Message-ID: <FB6881D5489143338D823F52AE004230@MOBILEE1705>

now I can see after changed in default config with wav :

/bin/cat: write error: Broken pipe
sh: line 1: 22006 Done(1)                 /bin/cat /tmp/mail.12687062250d3d
     22007 Segmentation fault      | sendmail -f 123456789 at mydomain -t owner at mydomain

however it seems that the file exists in /tmp but in a different subfolder and different name

/tmp/<mydomain>/<userid>/msg_99010fd0-f429-441f-960d-939bb545591c.wav

Amy idea ?


  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 8:31 PM
  Subject: Re: [Freeswitch-users] voicemail options


  You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.


  /b


  On Mar 15, 2010, at 7:21 PM, Madovsky wrote:


    Ok, for what reason ?






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dujinfang at gmail.com  Mon Mar 15 20:09:00 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 11:09:00 +0800
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
Message-ID: <23f91031003152009o7c67cb66ubdcba84bd41c040c@mail.gmail.com>

Using event consumer immediately after session:hangup

-- uuid = session.uuid
session:hangup
   for e in (function() return con:pop(1) end) do
             -- psudo code
             -- if e.uuid ==  uuid and e.event_type == reporting then
....
   end

note this is not safe since you may miss the event before you do
con:pop, then maybe you can try non-blocking con:pop() ...  not
tested, just an idea. Anyway, doing inline bill is not recommended.

2010/3/15 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> Hi All,
> I have the following simple Lua script answer a inbound call then dial
> another number and bridge the call.
> Is there any way to get the ?call duration?or billing sec using the
> session:getVariable( ?);
>
>
> -- answer the call
> session:answer();
> -- sleep a second
> session:sleep(1000);
> -- Initiate an outbound call
> obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> obSession:execute("sched_hangup","+60 alloted_timeout");
> -- bride the call
> ?? ?freeswitch.bridge(session, obSession);
> -- hangup
> session:hangup();
>
> Thank you
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Mon Mar 15 20:14:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 22:14:41 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <FB6881D5489143338D823F52AE004230@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
	<FB6881D5489143338D823F52AE004230@MOBILEE1705>
Message-ID: <47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>

Not too sure who Amy is... but did FreeSWITCH crash or your mailer?

/b

On Mar 15, 2010, at 9:26 PM, Madovsky wrote:

> Amy idea ?

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From infos at madovsky.org  Mon Mar 15 20:42:35 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 23:42:35 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705>
	<47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>
Message-ID: <5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>

fingers ripped...
no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

/bin/cat: write error: Broken pipe
sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
     15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

I think it means that the cat doesn't find the file...
so I checked the /tmp permission and it's ok (chmod 1777)
also if I change in voicemail.conf.xml 
<param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
FS continues to use /tmp.

Latest trunk, fedora 10 64bits
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Not too sure who Amy is... but did FreeSWITCH crash or your mailer?


  /b


  On Mar 15, 2010, at 9:26 PM, Madovsky wrote:


    Amy idea ?




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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 20:50:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 22:50:00 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705>
	<47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>
	<5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
Message-ID: <D4693B04-452E-453D-B9AF-71B1EC8F9EA6@freeswitch.org>

NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using?

/b

On Mar 15, 2010, at 10:42 PM, Madovsky wrote:

> fingers ripped...
> no FS is ok, I updated to 16998,  used the default voicemail config and it's the same
>  
> /bin/cat: write error: Broken pipe
> sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
>      15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba
>  
> I think it means that the cat doesn't find the file...
> so I checked the /tmp permission and it's ok (chmod 1777)
> also if I change in voicemail.conf.xml
> <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
> FS continues to use /tmp.
>  
> Latest trunk, fedora 10 64bits
> 

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From infos at madovsky.org  Mon Mar 15 20:58:30 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 23:58:30 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705><47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org><5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
	<D4693B04-452E-453D-B9AF-71B1EC8F9EA6@freeswitch.org>
Message-ID: <E6104FD0533A4D1B91F45A2AE37A1ED3@MOBILEE1705>

sendmail.
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:50 PM
  Subject: Re: [Freeswitch-users] voicemail options


  NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using?


  /b


  On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


    fingers ripped...
    no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

    /bin/cat: write error: Broken pipe
    sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
         15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

    I think it means that the cat doesn't find the file...
    so I checked the /tmp permission and it's ok (chmod 1777)
    also if I change in voicemail.conf.xml
    <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
    FS continues to use /tmp.

    Latest trunk, fedora 10 64bits






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 21:20:06 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:20:06 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>

So I checked t/mp WHILE the operator says "message saved"
but nothing like /tmp/mail.xxxxxxx

do you have an army of ideas ?
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:58 PM
  Subject: Re: [Freeswitch-users] voicemail options


  sendmail.
    ----- Original Message ----- 
    From: Brian West 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Monday, March 15, 2010 11:50 PM
    Subject: Re: [Freeswitch-users] voicemail options


    NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


    /b


    On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


      fingers ripped...
      no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

      /bin/cat: write error: Broken pipe
      sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
           15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

      I think it means that the cat doesn't find the file...
      so I checked the /tmp permission and it's ok (chmod 1777)
      also if I change in voicemail.conf.xml
      <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
      FS continues to use /tmp.

      Latest trunk, fedora 10 64bits






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


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 21:28:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:28:34 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>
References: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>
Message-ID: <90B89EA1-AF0B-4A4F-83C1-A5B208D86BE9@freeswitch.org>

Yes your mailer is crashing and never actually mailing the file... Works here for me.... I use exim see faq on wiki.

/b

On Mar 15, 2010, at 11:20 PM, Madovsky wrote:

> So I checked t/mp WHILE the operator says "message saved"
> but nothing like /tmp/mail.xxxxxxx
>  
> do you have an army of ideas ?

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From tayeb.meftah at gmail.com  Mon Mar 15 21:30:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:30:44 +0100
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <4B9F0974.70509@gmail.com>

yeah, this is right but svn up to 1.0.5!
you are 3yeers old
Le 15/03/2010 16:36, srinivasula reddy a ?crit :
> Hi,
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco 
> like just farwading the call.
> <action application="bridge" data= 
> <mailto:data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>sofia/internal/${destination_number}@ciscogatewayip 
> <mailto:sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>/>
> i do not using any gateway here, just farwarding call to router, is 
> this correct way of doing? any idea
>
> -- 
> Srinivasula Reddy K
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From tzury.by at reguluslabs.com  Mon Mar 15 21:31:28 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Tue, 16 Mar 2010 06:31:28 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com> 
	<c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
Message-ID: <10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>

On Mon, Mar 15, 2010 at 5:28 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Ok, so the call is not even being placed in the ISDN network.
>
> Did you manage to get the sangoma_prid logs working? if that's the case
> then paste them too, it seems sangoma_prid is not seeing the request from
> FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
> response from sangoma_prid binary (they both communicate through sctp
> socket).
>

This is the log (a result of `# /usr/sbin/sangoma_prid &> sangoma_prid.log`)

================System restart=============
= Sangoma PRI Protocol Stack Daemon       =
= Version: 1.58                           =
= Date: Feb 18 2010                       =
= Wanpipe Release: wanpipe-3.5.10.3          =
= Revision:Revision: 15364          =
===========================================
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.overload.task
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.100%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.90%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.overload.task
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.100%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.90%-full
WARN - detected an AFT-A101 board with an unsupported firmware version.
(detected: 0x34, supported: 0x36). You may encounter unknown issues if you
use an unsupported firmware version. Please contact support to get the
proper instructions to get the supported firmware unless you have been
instructed to use this firmware version.
ERROR - Assertion failed (in_eNewState == STARTING_TSLINK_STATE)
file=../../../stack/src/main.cpp line=134
ERROR - ACE_Task::wait failed with error 22 (Invalid argument) . Task name =
TelesoftStack
ERROR - failed to waitUntilDone the TaskManager's thread TelesoftStack
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From infos at madovsky.org  Mon Mar 15 21:33:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:33:58 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>

there's a thread here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
for exim with the same problem,
but at the end he didn't say if it worked or not
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:20 AM
  Subject: Re: [Freeswitch-users] voicemail options


  So I checked t/mp WHILE the operator says "message saved"
  but nothing like /tmp/mail.xxxxxxx

  do you have an army of ideas ?
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Monday, March 15, 2010 11:58 PM
    Subject: Re: [Freeswitch-users] voicemail options


    sendmail.
      ----- Original Message ----- 
      From: Brian West 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 11:50 PM
      Subject: Re: [Freeswitch-users] voicemail options


      NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


      /b


      On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


        fingers ripped...
        no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

        /bin/cat: write error: Broken pipe
        sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
             15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

        I think it means that the cat doesn't find the file...
        so I checked the /tmp permission and it's ok (chmod 1777)
        also if I change in voicemail.conf.xml
        <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
        FS continues to use /tmp.

        Latest trunk, fedora 10 64bits






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


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 21:38:21 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:38:21 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>
References: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>
Message-ID: <6C5774D8-D130-458F-BC40-0DC1F5ECE87B@freeswitch.org>

read the wiki it tells you how.

/b

On Mar 15, 2010, at 11:33 PM, Madovsky wrote:

> there's a thread here
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
> for exim with the same problem,
> but at the end he didn't say if it worked or not
> 

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From brian at freeswitch.org  Mon Mar 15 21:39:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:39:44 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <7E79BFF7-FC88-4333-AD25-370D55BB49AF@freeswitch.org>

You should NEVER and I mean NEVER use ${destination_number} as is... its possible to inject things into the dial string.  We never recommend it.

gun+bullets=shot in foot.

/b

On Mar 15, 2010, at 10:36 AM, srinivasula reddy wrote:

> <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>

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From tayeb.meftah at gmail.com  Mon Mar 15 21:44:53 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:44:53 +0100
Subject: [Freeswitch-users] latest windows binary
Message-ID: <4B9F0CC5.3020409@gmail.com>

hi all,
could anyone build a windows version and put it in latest.freeswitch.org?
thanks


From infos at madovsky.org  Mon Mar 15 21:46:41 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:46:41 -0400
Subject: [Freeswitch-users] voicemail options [SOLVED]
Message-ID: <AD69B0257309443D943A3F48754B9C00@MOBILEE1705>

as my sendmail is ultra secure I forgot that the FROM "email"
MUST be registered in your mails server...

but now there is another problem. I open a new thread

Thanks for your time

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:33 AM
  Subject: Re: [Freeswitch-users] voicemail options


  there's a thread here
  http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
  for exim with the same problem,
  but at the end he didn't say if it worked or not
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 12:20 AM
    Subject: Re: [Freeswitch-users] voicemail options


    So I checked t/mp WHILE the operator says "message saved"
    but nothing like /tmp/mail.xxxxxxx

    do you have an army of ideas ?
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 11:58 PM
      Subject: Re: [Freeswitch-users] voicemail options


      sendmail.
        ----- Original Message ----- 
        From: Brian West 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Monday, March 15, 2010 11:50 PM
        Subject: Re: [Freeswitch-users] voicemail options


        NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


        /b


        On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


          fingers ripped...
          no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

          /bin/cat: write error: Broken pipe
          sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
               15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

          I think it means that the cat doesn't find the file...
          so I checked the /tmp permission and it's ok (chmod 1777)
          also if I change in voicemail.conf.xml
          <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
          FS continues to use /tmp.

          Latest trunk, fedora 10 64bits






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


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org
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From infos at madovsky.org  Mon Mar 15 21:49:33 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:49:33 -0400
Subject: [Freeswitch-users] voicemail message "saved" error
Message-ID: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>

Hi,

2010-03-16 00:44:27.878289 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns
sh: line 1: 30667 Done                    /bin/cat /tmp/mail.126871466794cf
     30668 Segmentation fault      | sendmail -f infos at bibi.baba -t other at kiki.coco


Thanks

F
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From tayeb.meftah at gmail.com  Mon Mar 15 21:55:10 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:55:10 +0100
Subject: [Freeswitch-users] PSTN connection
In-Reply-To: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>
References: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>
Message-ID: <4B9F0F2E.4060407@gmail.com>

hi,
mayb you can wait for mod_gsmopen;)
i am not sure.
Le 11/03/2010 18:34, Ken Gillett a ?crit :
> If I am running FreeSwitch on a Mac, would I be able to use Apple's USB modem as a means to connect to the PSTN and utilise this line to place and receive calls?
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From tayeb.meftah at gmail.com  Mon Mar 15 21:56:39 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:56:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>	<4B951E63.6060200@coppice.org>	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>	<191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
	<65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>
Message-ID: <4B9F0F87.1050001@gmail.com>

mod_fax is ready andd in production no T38 only T30

Le 11/03/2010 18:15, Tihomir Culjaga a ?crit :
>
>
> On Wed, Mar 10, 2010 at 6:11 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     We have a mod_fax for t30 and work on t38 is underway.
>     So we already acknowledge that I guess.
>
>
> anything ready for testing ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From infos at madovsky.org  Mon Mar 15 21:57:55 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:57:55 -0400
Subject: [Freeswitch-users] voicemail options [SOLVED]
Message-ID: <F2B4DA51682D4F8E963BF7A657F56FA0@MOBILEE1705>

Brian, I read the wiki 2 days ago already and din't help me
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:46 AM
  Subject: Re: [Freeswitch-users] voicemail options [SOLVED]


  as my sendmail is ultra secure I forgot that the FROM "email"
  MUST be registered in your mails server...

  but now there is another problem. I open a new thread

  Thanks for your time

  Franck
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 12:33 AM
    Subject: Re: [Freeswitch-users] voicemail options


    there's a thread here
    http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
    for exim with the same problem,
    but at the end he didn't say if it worked or not
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 16, 2010 12:20 AM
      Subject: Re: [Freeswitch-users] voicemail options


      So I checked t/mp WHILE the operator says "message saved"
      but nothing like /tmp/mail.xxxxxxx

      do you have an army of ideas ?
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Monday, March 15, 2010 11:58 PM
        Subject: Re: [Freeswitch-users] voicemail options


        sendmail.
          ----- Original Message ----- 
          From: Brian West 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 11:50 PM
          Subject: Re: [Freeswitch-users] voicemail options


          NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


          /b


          On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


            fingers ripped...
            no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

            /bin/cat: write error: Broken pipe
            sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
                 15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

            I think it means that the cat doesn't find the file...
            so I checked the /tmp permission and it's ok (chmod 1777)
            also if I change in voicemail.conf.xml
            <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
            FS continues to use /tmp.

            Latest trunk, fedora 10 64bits






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


          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tayeb.meftah at gmail.com  Mon Mar 15 22:01:52 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 06:01:52 +0100
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
Message-ID: <4B9F10C0.6020501@gmail.com>

hi
don't confuse betwan internal/external profile
external profile is to send trafic to a provider and to subscrib users
but in internal you can subscrib external and internal users awell
just forward the required ports like 5060TCP/UDP for sip and 16000-32000 
for rtp

Le 11/03/2010 01:41, Vladimir Elizarov a ?crit :
> Hello.
>
> What are the ways connect external users to the fs there? In such a 
> scheme:
> fs with real ip <-> internet <-> nat <-> sip-phone
>
>
> I read the topic http://wiki.freeswitch.org/wiki/External_profile # 
> Switch_with_External_SoftPhone
>
> I did not understand several things:
> - That should be in exteranl5090.xml? Whole profile external.xml c 
> changes in paragraphs 2,3?
> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes 
> in xxxx? External UA IP? Domain?" I have the same question. that 
> x.x.x.x should be?
>
> But this seems not quite what I need. Must be external to the 
> subscriber is practically no different from internal.
>
> -- 
> Best regards, Vladimir Elizarov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From brian at freeswitch.org  Mon Mar 15 22:08:22 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 00:08:22 -0500
Subject: [Freeswitch-users] voicemail message "saved" error
In-Reply-To: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>
References: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>
Message-ID: <4E102168-434C-4DC7-9D38-661D890BD056@freeswitch.org>

make vm-sync
restart

On Mar 15, 2010, at 11:49 PM, Madovsky wrote:

> 2010-03-16 00:44:27.878289 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns
> sh: line 1: 30667 Done                    /bin/cat /tmp/mail.126871466794cf
>      30668 Segmentation fault      | sendmail -f infos at bibi.baba -t other at kiki.coco
>  
> Thanks
>  

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From ahmed.ajmal at breezecom.ae  Mon Mar 15 22:41:41 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Tue, 16 Mar 2010 10:41:41 +0500
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>	<8259813661645760166@unknownmsgid>
	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
Message-ID: <015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>

I am looking at the FS console after hangup is complete. I also have a ESL
event hooked up for HANGUP_COMPLETE event. In both cases I am unable to see
the sip_gateway_name variable.

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 15, 2010 11:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

 

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>
wrote:

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Where are you looking to find the variable?
-MC

 

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From ahmed.ajmal at breezecom.ae  Mon Mar 15 23:21:49 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Tue, 16 Mar 2010 11:21:49 +0500
Subject: [Freeswitch-users] Set/Export channel variable [SOLVED]
In-Reply-To: <015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>	<8259813661645760166@unknownmsgid>	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
	<015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>
Message-ID: <016701cac4d0$f917d210$eb477630$@ajmal@breezecom.ae>

So it seems that the FS console by default only displays the 'a' leg vars
which is why I couldn't see it (and it makes sense too since gateway is on
the b leg side). I have the ESL script now display both the inbound/outbound
after which I can see the sip_gateway_name var. 

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ahmed
Ajmal
Sent: Tuesday, March 16, 2010 10:42 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

I am looking at the FS console after hangup is complete. I also have a ESL
event hooked up for HANGUP_COMPLETE event. In both cases I am unable to see
the sip_gateway_name variable.

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 15, 2010 11:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

 

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>
wrote:

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Where are you looking to find the variable?
-MC

 

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From infos at madovsky.org  Mon Mar 15 23:32:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 02:32:10 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
Message-ID: <1B3CCF4C5CD84CF28FA7270D0C20EA2B@MOBILEE1705>

Sorry Brian, I just realised that
the wav is encoded in 8khz for vociemail.
So it's not  big MB deal

Good Night
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 5:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Word of advice.  MP3 is not recommended at all.


  /b


  On Mar 15, 2010, at 3:59 PM, Madovsky wrote:


    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tayeb.meftah at gmail.com  Tue Mar 16 01:00:23 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 09:00:23 +0100
Subject: [Freeswitch-users] answering call in irly_media mode
Message-ID: <4B9F3A97.4030609@gmail.com>

hi all
i want to setup a custom ringback tone for inbound (not outbound calls)
is in irly_media mode but i still heare the normal US ringback.
where is my dialplan extension:
http://pastebin.freeswitch.org/12452
thanks


From will.traenkle at yahoo.com  Tue Mar 16 01:03:32 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 16 Mar 2010 01:03:32 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
	<923176.22123.qm@web57601.mail.re1.yahoo.com>
	<87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
Message-ID: <617776.63524.qm@web57601.mail.re1.yahoo.com>

Michael,

Thanks for your help.  I was able to get it work, however, I would like your opinion as how to do this correctly as I have a hunch I may have opened up some security risks or have gone against freeswitch best practices.  If not, just let me know and I will leave it as is.

I modified the public.xml file under /conf/dialplan/


I uncommented the following:

<!-- TQL
I removed this because of the auto attendant to hunt group to external number error.
I am sure there is some security or other issue in doing this.
But hey, at least my hunt groups work.

    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>
-->

I added the following:

<!-- TQL
I added this to pick up calls being sent to voicemail.
Note, put this after public_extensions
-->

    <extension name="send_to_voicemail">
      <condition field="destination_number" expression="^\*99(\d{3,6})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

<!-- TQL
I added this to pick up the hunt groups. 
Note, put this after the public/*.xml includes
-->

        <X-PRE-PROCESS cmd="include" data="default/*.xml"/>






________________________________
From: Michael Collins <msc at freeswitch.org>
To: William Traenkle <will.traenkle at yahoo.com>
Sent: Mon, March 15, 2010 5:11:57 PM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds




On Mon, Mar 15, 2010 at 4:22 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials
>an internal extension via correctly but when trying to dial an
>external number such a cell phone via sip uri it is unable to connect
>or ring the cell phone and it waits the correct timeout period before it
>goes to voicemail.
>
>However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  
>
>
>
>>There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.
>
>>1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR
>
>>2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3
In #2 your call to 7000 is hitting the public context. Most likely this isn't what you were anticipating. Try handling 7000 in the public context. Find the public_extensions dp entry and update your regex there:
<condition field="destination_number" expression="^(10[01][0-9]|7000)$">

reloadxml and try again. If it works then consider creating a public context extension that specifically handles these calls, perhaps checking the IP address of the source of the call. 
-MC


>
>>3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3
>
>>4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
>
> 
>5) Dialplan default.xml:  
>http://pastebin.com/iAbNSW9E 
> 
>If you require additional information or have any other
>questions, please let me know.
>
>>Thanks,
>
>>-Will
>
________________________________
 From: Michael Collins <msc at freeswitch.org>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tue, March 9, 2010 11:00:53 PM
>Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
>
>
>
>
>On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:
>
>When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>>
>>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>>
>>Any ideas where I should look and what line of code I might be missing?
>>
>>
>
>Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
>>
>-MC
>
>
>



      
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From will.traenkle at yahoo.com  Tue Mar 16 01:09:53 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 16 Mar 2010 01:09:53 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
Message-ID: <227536.63049.qm@web57605.mail.re1.yahoo.com>

Michael,

Last question, what dictates when a call uses the public.xml or the default.xml dial plan?

Thanks,





________________________________
From: William Traenkle <will.traenkle at yahoo.com>
To: Michael Collins <msc at freeswitch.org>; freeswitch-users at lists.freeswitch.org
Sent: Tue, March 16, 2010 1:03:32 AM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds


Michael,

Thanks for your help.  I was able to get it work, however, I would like your opinion as how to do this correctly as I have a hunch I may have opened up some security risks or have gone against freeswitch best practices.  If not, just let me know and I will leave it as is.

I modified the public.xml file under /conf/dialplan/


I uncommented the following:

<!-- TQL
I removed this because of the auto attendant to hunt group to external number error.
I am sure there is some security or other issue in doing this.
But hey, at least my hunt groups work.

    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>
-->

I added the following:

<!-- TQL
I added this to pick up calls being sent to voicemail.
Note, put this after public_extensions
-->

    <extension name="send_to_voicemail">
      <condition field="destination_number" expression="^\*99(\d{3,6})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

<!-- TQL
I added this to pick up the hunt groups. 
Note, put this after the public/*.xml includes
-->

        <X-PRE-PROCESS cmd="include" data="default/*.xml"/>






________________________________
From: Michael Collins <msc at freeswitch.org>
To: William Traenkle <will.traenkle at yahoo.com>
Sent: Mon, March 15, 2010 5:11:57 PM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds




On Mon, Mar 15, 2010 at 4:22 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials
>an internal extension via correctly but when trying to dial an
>external number such a cell phone via sip uri it is unable to connect
>or ring the cell phone and it waits the correct timeout period before it
>goes to voicemail.
>
>However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  
>
>
>
>>There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.
>
>>1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR
>
>>2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3
In #2 your call to 7000 is hitting the public context. Most likely this isn't what you were anticipating. Try handling 7000 in the public context. Find the public_extensions dp entry and update your regex there:
<condition field="destination_number" expression="^(10[01][0-9]|7000)$">

reloadxml and try again. If it works then consider creating a public context extension that specifically handles these calls, perhaps checking the IP address of the source of the call. 
-MC


>
>>3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3
>
>>4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
>
> 
>5) Dialplan default.xml:  
>http://pastebin.com/iAbNSW9E 
> 
>If you require additional information or have any other
>questions, please let me know.
>
>>Thanks,
>
>>-Will
>
________________________________
 From: Michael Collins <msc at freeswitch.org>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tue, March 9, 2010 11:00:53 PM
>Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
>
>
>
>
>On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:
>
>When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>>
>>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>>
>>Any ideas where I should look and what line of code I might be missing?
>>
>>
>
>Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
>>
>-MC
>
>
>


      
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From jason at jasonjgw.net  Tue Mar 16 01:34:58 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Mar 2010 19:34:58 +1100
Subject: [Freeswitch-users] IVR to hunt group to external destination
 fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <227536.63049.qm@web57605.mail.re1.yahoo.com>
References: <227536.63049.qm@web57605.mail.re1.yahoo.com>
Message-ID: <20100316083458.GA30096@jdc.jasonjgw.net>

William Traenkle <will.traenkle at yahoo.com> wrote:
> 
> Last question, what dictates when a call uses the public.xml or the
> default.xml dial plan?

The context of the call, e.g., as specified in the SIP profile.



From jason at jasonjgw.net  Tue Mar 16 01:45:09 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Mar 2010 19:45:09 +1100
Subject: [Freeswitch-users] IVR to hunt group to external destination
 fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <617776.63524.qm@web57601.mail.re1.yahoo.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
	<923176.22123.qm@web57601.mail.re1.yahoo.com>
	<87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
	<617776.63524.qm@web57601.mail.re1.yahoo.com>
Message-ID: <20100316084509.GA30197@jdc.jasonjgw.net>

William Traenkle <will.traenkle at yahoo.com> wrote:
> Thanks for your help.  I was able to get it work, however, I would like your
> opinion as how to do this correctly as I have a hunch I may have opened up
> some security risks or have gone against freeswitch best practices.  If not,
> just let me know and I will leave it as is.

What you don't want is for calls originating in the public context to be
transferred into extensions of the default dial plan for which they aren't
authorized.

Based on a cursory reading of your post, I don't notice any problems; you
aren't transferring the calls into the default dial plan using the transfer
application, so it should all just work fine.

Having said that, I'm not responsible for your security; it's your task to
consider call flows and make sure that public calls won't end up where you
don't want them.



From dujinfang at gmail.com  Tue Mar 16 02:53:21 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 17:53:21 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
Message-ID: <23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> FS would prefer you run your kernel at 1000hz you are probably on a VM if
> its 10000hz
>

It's a  physical machine. And I'm also running 20 skype clients. as
100HZ is recommended for skypiax. I think the kernel should be 100 or
250 HZ. weird.


here is the dingaling log with dl_debug on

http://pastebin.freeswitch.org/12453
My public IP is 220.231.26.130, and nat to 192.168.1.27(me)


still get this message on client:

Sorry! The voice chat with language_lesson failed because of a network
problem at 5:39 PM. Please try again.
 Click here to report this to Google



Thanks.









>
> On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> Windows client and web site on Mac. They both work on the other two
>> instances of FreeSWITCH I mentioned in this thread.
>>
>> btw, can you give me a hint on the timer warning?
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > you would have to execute
>> >
>> > dl_debug on
>> >
>> > to get a trace that would tell us anything about that?
>> >
>> > Are you using the windows gtalk client or the web site?
>> > We are following google who have been pushing for the online client
>> > embedded
>> > in the gmail website.
>> >
>> >
>> >
>> >
>> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> here is the client:
>> >>
>> >> <profile type="client">
>> >> ?<param name="name" value="gmail.com"/>
>> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> ?<param name="password" value="password"/>
>> >> ?<param name="server" value="talk.google.com"/>
>> >> ?<param name="dialplan" value="XML"/>
>> >> ?<param name="contex" value="public"/>
>> >> ?<param name="message" value="FreeSWITCH"/>
>> >> ?<param name="rtp-ip" value="auto"/>
>> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> ?<param name="auto-login" value="true"/>
>> >> ?<param name="auto-reply" value="Welcome"/>
>> >> ?<param name="sasl" value="plain"/>
>> >> ?<param name="tls" value="true"/>
>> >> ?<param name="exten" value="5000"/>
>> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> </profile>
>> >>
>> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> > ALL,
>> >> >
>> >> > I just upgrade FS to 16992.
>> >> >
>> >> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
>> >> > ok.
>> >> >
>> >> >
>> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> >> > detected!
>> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> > experience audio problems.
>> >> >
>> >> > uname -a
>> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
>> >> > 2008 i686 GNU/Linux
>> >> >
>> >> >
>> >> > I used a script to check the interupts
>> >> >
>> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >
>> >> >
>> >> > 2) run dingaling with client mode. dingaling doesn't pick up answer.
>> >> > I
>> >> > originate a call from FS, the console blocks, gtalk client ring and
>> >> > it
>> >> > looks ok when I hit answer, but no sound, FS console still blocks. a
>> >> > few seconds later gtalk client shows:
>> >> >
>> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> > network
>> >> > problem at 5:50 PM. Please try again.
>> >> > ?Click here to report this to Google.
>> >> >
>> >> >
>> >> > here is a log with bgapi:
>> >> > http://pastebin.freeswitch.org/12434
>> >> >
>> >> >
>> >> > rev 14696 works on the same server same conf. another server running
>> >> > 16958M works at the same time. I have no clue to check, is it related
>> >> > to the timer as in 1)? can some help me to take a look? let me know
>> >> > if
>> >> > you need "dl_debug on" logs.
>> >> >
>> >> > Thanks.
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jbrucehopkins at gmail.com  Tue Mar 16 03:30:43 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 10:30:43 +0000
Subject: [Freeswitch-users] Speex transcoding question
Message-ID: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>

Hi,

I am having trouble getting my configuration  right so that I can have a
call transcoded to Speex wideband from another codec (alaw or g.722).

If both phones use Speex wideband with no transcoding required by FS, the
call succeeds though.

My codecs are listed in vars.xml as follows:
 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
@20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
@20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

However if I make a call from a phone A using, say, g.722 to a phone using
Speex wideband, the SDP in the invite from phone A to phone B does not
include Speex wideband.  In fact the SDP includes speex/8000 even though
speex/8000 is neither enabled in vars.xml, not in either of the phones.
Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
B (from Wireshark) :

Media Attribute (a): rtpmap:9 G722/8000
Media Attribute (a): rtpmap:99 SPEEX/8000
Media Attribute (a): rtpmap:115 G7221/32000
Media Attribute (a): fmtp:115 bitrate=48000
Media Attribute (a): rtpmap:107 G7221/16000
Media Attribute (a): fmtp:107 bitrate=32000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): rtpmap:13 CN/8000
Media Attribute (a): ptime:20

Could anyone tell me what I am doing wrong please?

Many thanks
Bruce
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From jbrucehopkins at gmail.com  Tue Mar 16 03:36:01 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 10:36:01 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
Message-ID: <d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>

Please ignore this last email.  I think I have missed out something I hav
already been told.  Sorry for the waste of bandwidth.

regards
Bruce

On 16 March 2010 10:30, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi,
>
> I am having trouble getting my configuration  right so that I can have a
> call transcoded to Speex wideband from another codec (alaw or g.722).
>
> If both phones use Speex wideband with no transcoding required by FS, the
> call succeeds though.
>
> My codecs are listed in vars.xml as follows:
>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>
> However if I make a call from a phone A using, say, g.722 to a phone using
> Speex wideband, the SDP in the invite from phone A to phone B does not
> include Speex wideband.  In fact the SDP includes speex/8000 even though
> speex/8000 is neither enabled in vars.xml, not in either of the phones.
> Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
> B (from Wireshark) :
>
> Media Attribute (a): rtpmap:9 G722/8000
> Media Attribute (a): rtpmap:99 SPEEX/8000
> Media Attribute (a): rtpmap:115 G7221/32000
> Media Attribute (a): fmtp:115 bitrate=48000
> Media Attribute (a): rtpmap:107 G7221/16000
> Media Attribute (a): fmtp:107 bitrate=32000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:3 GSM/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
> Media Attribute (a): rtpmap:13 CN/8000
> Media Attribute (a): ptime:20
>
> Could anyone tell me what I am doing wrong please?
>
> Many thanks
> Bruce
>
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From ken at ukgb.net  Tue Mar 16 03:50:05 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 10:50:05 +0000
Subject: [Freeswitch-users] call info display
Message-ID: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>

Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.

When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).

Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?

So, how does FS handle all the above?
What does it pass to the recipient, for both external and internal calls?
Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?


Ken G i l l e t t

_/_/_/_/_/_/_/_/



From siniypin at gmail.com  Tue Mar 16 04:18:55 2010
From: siniypin at gmail.com (RobertT)
Date: Tue, 16 Mar 2010 14:18:55 +0300
Subject: [Freeswitch-users] ICE related sdp headers forwarding
In-Reply-To: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>
References: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>
Message-ID: <2160023e1003160418m163f1badq4fea3df25456fe04@mail.gmail.com>

Should someone want to know the answers I found, here they are.

I had two ideas about how to forward SDP ICE-related headers from FS to
leg-B.
*1. Redirect a directly to b. But to accomplish this I've to know b's IP,
that it was registered with, which is being stored in sip_registrations
db. Can I obtain it with db app in dialplan and substitute as a parameter to
deflect app?
*This is* *rather simple - just call "defend" app from dialplan with
${destination_number} parameter. Actually, FS answers with a bit awkward
address in Refer-to header, containing <profile_name><destination>@<domain>,
so one would have to parse recieved address manually, to use a correct sip
uri. Did I make something wrong?..
*
*
*2. Force FS to copy remote SDP headers, or at least a part of them, to it's
local SDP.*
This is even more simple - just set inbound_bypass_media to true in sofia
profile settings, and FS will leave all of the SDP headers intact. But, you
have to remember, that media traffic for this profile will bypass FS, as
parameter's name implies. Therefore FS will not be able to manipulate media
data in anyway.

Best regards,
RobertT.
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From jbrucehopkins at gmail.com  Tue Mar 16 04:26:21 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 11:26:21 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
Message-ID: <d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>

OK - sorry to make such a meal of this.

I am still encountering a problem getting my configuration right, having
followed the advice I received previously.

What I am observing is that if more than one flavour of SPEEX is enabled in
vars.xml, then only the first one in the list is offered to phone B in the
SDP section of the INVITE if transcoding is required by Freeswitch.

i.e. In vars.xml I have:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=SPEEX at 16000h,SPEEX at 32000h
,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

 <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=SPEEX at 16000h
,SPEEX at 32000h,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>


If I then try to make a call from Phone A (using PCMA) to Phone B (using
SPEEX/8000), the call fails and the log shows an incompatible destination.
The SDP of the INVITE from Freeswitch to Phone B offers SPEEX/16000 but not
SPEEX/8000 or SPEEX/32000, so there is no compatible codec offered to Phone
B.

If however both Phone A and Phone B both use SPEEX/8000, the call works
fine  - as my default configuration is that the codec preferred by Phone A
goes to the top of the list of codecs offered to Phone B

Could anyone help with what I need to do to resolve my problem when
transcoding is required though please?

I am did "make current" last night so am using the latest snapshop.

many thanks again,
Bruce





On 16 March 2010 10:36, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Please ignore this last email.  I think I have missed out something I hav
> already been told.  Sorry for the waste of bandwidth.
>
> regards
> Bruce
>
>
> On 16 March 2010 10:30, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi,
>>
>> I am having trouble getting my configuration  right so that I can have a
>> call transcoded to Speex wideband from another codec (alaw or g.722).
>>
>> If both phones use Speex wideband with no transcoding required by FS, the
>> call succeeds though.
>>
>> My codecs are listed in vars.xml as follows:
>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> However if I make a call from a phone A using, say, g.722 to a phone using
>> Speex wideband, the SDP in the invite from phone A to phone B does not
>> include Speex wideband.  In fact the SDP includes speex/8000 even though
>> speex/8000 is neither enabled in vars.xml, not in either of the phones.
>> Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
>> B (from Wireshark) :
>>
>> Media Attribute (a): rtpmap:9 G722/8000
>> Media Attribute (a): rtpmap:99 SPEEX/8000
>> Media Attribute (a): rtpmap:115 G7221/32000
>> Media Attribute (a): fmtp:115 bitrate=48000
>> Media Attribute (a): rtpmap:107 G7221/16000
>> Media Attribute (a): fmtp:107 bitrate=32000
>> Media Attribute (a): rtpmap:8 PCMA/8000
>> Media Attribute (a): rtpmap:0 PCMU/8000
>> Media Attribute (a): rtpmap:3 GSM/8000
>> Media Attribute (a): rtpmap:101 telephone-event/8000
>> Media Attribute (a): fmtp:101 0-16
>> Media Attribute (a): rtpmap:13 CN/8000
>> Media Attribute (a): ptime:20
>>
>> Could anyone tell me what I am doing wrong please?
>>
>> Many thanks
>> Bruce
>>
>
>
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From peter.olsson at visionutveckling.se  Tue Mar 16 04:41:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 12:41:09 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>

I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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

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From Prometheus001 at gmx.net  Tue Mar 16 04:55:25 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Mar 2010 12:55:25 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
Message-ID: <4B9F71AD.7070008@gmx.net>

I have the folowing scenario:
  A Snom phone has activated DND (do not disturb)
  When the phone is called, it returns "480 Do Not Disturb" and then
"404 Not Found".
  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
  The PSTN then (wrongly) announces "this number is not listed".

My question: How can we overcome this and announce back a different
message to PSTN, but only on NO_USER_RESPONSE.
I tried to continue on bridge and then this after the bridge command the
following:
      <condition field="variable_originate_disposition"
expression="NO_USER_RESPONSE" inline="true">
        <action application="hangup" data="USER_BUSY"/>
      </condition>

But the installed Freeswith does not support the "inline" yet and we
will not be able to update Freeswitch soon.

So my question:
Is there any other way to achieve this?

Best regards
Peter



From vetali100 at gmail.com  Tue Mar 16 05:04:21 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 14:04:21 +0200
Subject: [Freeswitch-users]  Load balancing solution using pure FreeSWITCH
Message-ID: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>

Hi dear FS community,

After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109)
and (
http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
use MySQL).

It works on few of our servers right now without any issues.

Feel free to reuse / improve it or provide your comments if you see
something wrong with it.


We have one routing server (RS) and several Media Servers (MS).
FreeSWITCH is installed on each server in default configuration.

--> MS_1
RS  --> MS_2
--> MS_N


(Failover can be implemented by adding one more routing server, but this is
out of scope of current description).

Routing server routes calls to media servers based on logic that you can
easily define.
(Generally speaking, Routing server can route part of the calls onto itself,
but to simplify let's say that it will route only to Media Servers).


Routing server is used to register subscribers (it will have directory files
./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
Media servers do NOT contain files with users (security issue? probably
should be managed on dialplan stage of media servers by comparing ip
addresses of requests).


The whole routing can be presented by the following chain:

Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
"sofia/external/" to determined MS IP address-->[Bridge to fail over media
server, optionally]-->End

Let's say we need to route all international calls based on less loaded
server (other logic can be easily implemented in PHP script as well).

*1. RS Dialplan: default.xml*
<include>
        <extension name="route">
                <condition field="destination_number"
expression="^(00)(\d+)">
                        <action application="lua" data="route.lua $2"/>
                </condition>
        </extension>
</include>


*2. RS Lua script: route.lua*
called_number = argv[1]; //without "00"
api = freeswitch.API();
server_ip_address = api:executeString("curl
http://localhost/get_server_ip.php);
forwarding_session_string =
"sofia/external/"..called_number.."@"..server_ip_address;
session:setVariable("bypass_media", "true"); //described below!!!
session:execute("bridge",forwarding_session_string);

*3. PHP script: get_server_ip.php*
<<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
LOADED AT THE MOMENT SERVER>>>
die($less_loaded_server_ip);


All Media Servers should provide information about their current load into
one common database (dedicated server or routing server can be used for
database).
If you use scripts you can update this table on starting a call and after
call finished.
Or you can create a job that will query FreeSWITCH and will update the
database every, say, 10 seconds.


Basically that's all (really!)
I beleive it can be used for most general purpose scenarios without need of
third-party tools.

P.S.
The only problem we had from the beginning was that RTP media traffic was
passing thru both servers: RS and Media Server.
But after we added the following command to the routing script, RTP media
traffic started to pass only thru Media server, making RS almost free.
session:setVariable("bypass_media", "true");



Best regards,
vIT
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From leon at scarlet-internet.nl  Tue Mar 16 05:42:56 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 16 Mar 2010 13:42:56 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <4B9F71AD.7070008@gmx.net>
References: <4B9F71AD.7070008@gmx.net>
Message-ID: <2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>

Hi,

Don't know if it helps, but the inline attribute can only be set on  
actions, not on conditions.

Also, the hangup application is not allowed to be run inline:
http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_dptools/mod_dptools.c?r=HEAD#l3047
(it has no SAF_ROUTING_EXEC flag)

Perhaps you can try to transfer to another context directly after your  
bridge ? (where that second context contains an extension with  
conditions matching on variable_originate_disposition)

kind regards,

Leon


On Mar 16, 2010, at 12:55 PM, Peter P GMX wrote:

> I have the folowing scenario:
>  A Snom phone has activated DND (do not disturb)
>  When the phone is called, it returns "480 Do Not Disturb" and then
> "404 Not Found".
>  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
>  The PSTN then (wrongly) announces "this number is not listed".
>
> My question: How can we overcome this and announce back a different
> message to PSTN, but only on NO_USER_RESPONSE.
> I tried to continue on bridge and then this after the bridge command  
> the
> following:
>      <condition field="variable_originate_disposition"
> expression="NO_USER_RESPONSE" inline="true">
>        <action application="hangup" data="USER_BUSY"/>
>      </condition>
>
> But the installed Freeswith does not support the "inline" yet and we
> will not be able to update Freeswitch soon.
>
> So my question:
> Is there any other way to achieve this?
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tculjaga at gmail.com  Tue Mar 16 06:07:39 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 14:07:39 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
Message-ID: <65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>

this is something you took from Mera .. isn't it ? :)



On Tue, Mar 16, 2010 at 1:04 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi dear FS community,
>
> After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109) and (
> http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
> solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
> use MySQL).
>
> It works on few of our servers right now without any issues.
>
> Feel free to reuse / improve it or provide your comments if you see
> something wrong with it.
>
>
> We have one routing server (RS) and several Media Servers (MS).
> FreeSWITCH is installed on each server in default configuration.
>
> --> MS_1
> RS  --> MS_2
> --> MS_N
>
>
> (Failover can be implemented by adding one more routing server, but this is
> out of scope of current description).
>

routing server is your single point of failure, you should make two of them
in active/standby mode. What about registered endpoints ? ... where do you
register them?.... you should have a DB replication in order to have the
"registered context" distributed between both routing servers


So, i'd suggest to bring up a LAN_HA and have a floating address between
your routing servers.... of course the you should be using ODBC in the core
to have registrations on both servers.




>
> Routing server routes calls to media servers based on logic that you can
> easily define.
> (Generally speaking, Routing server can route part of the calls onto
> itself, but to simplify let's say that it will route only to Media Servers).
>
> you will be loosing calls on the on the MS if it goes down ... maybe there
is a way to "replace" media resource for that calls and keep them going even
after the MS went down.



>
> Routing server is used to register subscribers (it will have directory
> files
> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> Media servers do NOT contain files with users (security issue? probably
> should be managed on dialplan stage of media servers by comparing ip
> addresses of requests).
>
>
consider ODBC in the core!


>
> The whole routing can be presented by the following chain:
>
> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> "sofia/external/" to determined MS IP address-->[Bridge to fail over media
> server, optionally]-->End
>
> Let's say we need to route all international calls based on less loaded
> server (other logic can be easily implemented in PHP script as well).
>
> *1. RS Dialplan: default.xml*
> <include>
>         <extension name="route">
>                 <condition field="destination_number"
> expression="^(00)(\d+)">
>                         <action application="lua" data="route.lua $2"/>
>                 </condition>
>         </extension>
> </include>
>
>
> *2. RS Lua script: route.lua*
> called_number = argv[1]; //without "00"
> api = freeswitch.API();
> server_ip_address = api:executeString("curl
> http://localhost/get_server_ip.php);
> forwarding_session_string =
> "sofia/external/"..called_number.."@"..server_ip_address;
> session:setVariable("bypass_media", "true"); //described below!!!
> session:execute("bridge",forwarding_session_string);
>
>
do your bridging in the dialplan not from a script... you will get better
performances!




> *3. PHP script: get_server_ip.php*
> <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> LOADED AT THE MOMENT SERVER>>>
> die($less_loaded_server_ip);
>
>
> All Media Servers should provide information about their current load into
> one common database (dedicated server or routing server can be used for
> database).
> If you use scripts you can update this table on starting a call and after
> call finished.
> Or you can create a job that will query FreeSWITCH and will update the
> database every, say, 10 seconds.
>
>
> Basically that's all (really!)
> I beleive it can be used for most general purpose scenarios without need of
> third-party tools.
>
> P.S.
> The only problem we had from the beginning was that RTP media traffic was
> passing thru both servers: RS and Media Server.
> But after we added the following command to the routing script, RTP media
> traffic started to pass only thru Media server, making RS almost free.
> session:setVariable("bypass_media", "true");
>
>
>
anyhow it is a good start ... with a custom module for media server
"kepalive" and resource migration it could be a great solution.


T.
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From lawwton at gmail.com  Tue Mar 16 06:13:14 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Tue, 16 Mar 2010 09:13:14 -0400
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
Message-ID: <5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>

Vit:

Nice email, very informative; curious to find out the reasons why you
decided to go with FS instead of OpenSip. We soon have to make a
similar decision and would like to find out the rationale behind it.

Thanks,

Alfredo

On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:
> Hi dear FS community,
> After a brief analysis of 2 solutions using OpenSIPS
> and?Kamailio?(http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
> ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my
> own solution using ONLY FreeSWITCH and ONLY 1 table of any database server
> (we use MySQL).
> It works on few of our servers right now without any issues.
> Feel free to reuse / improve it or provide your comments if you see
> something wrong with it.
>
> We have one routing server (RS) and several Media Servers (MS).
> FreeSWITCH is installed on each server in default configuration.
> --> MS_1
> RS ?--> MS_2
> --> MS_N
>
> (Failover can be implemented by adding one more routing server, but this is
> out of scope of current description).
> Routing server routes calls to media servers based on logic that you can
> easily define.
> (Generally speaking, Routing server can route part of the calls onto itself,
> but to simplify let's say that it will route only to Media Servers).
>
> Routing server is used to register subscribers (it will have directory files
> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> Media servers do NOT contain files with users (security issue? probably
> should be managed on dialplan stage of media servers by comparing ip
> addresses of requests).
>
> The whole routing can be presented by the following chain:
> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> "sofia/external/"?to determined MS IP address-->[Bridge to fail over media
> server, optionally]-->End
> Let's say we need to route all international calls based on less loaded
> server (other logic can be easily implemented in PHP script as well).
> 1. RS Dialplan: default.xml
> <include>
> ?? ? ? ?<extension name="route">
> ?? ? ? ? ? ? ? ?<condition field="destination_number"
> expression="^(00)(\d+)">
> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="lua" data="route.lua $2"/>
> ?? ? ? ? ? ? ? ?</condition>
> ?? ? ? ?</extension>
> </include>
>
> 2. RS Lua script: route.lua
> called_number = argv[1]; //without "00"
> api = freeswitch.API();
> server_ip_address = api:executeString("curl
> http://localhost/get_server_ip.php);
> forwarding_session_string =
> "sofia/external/"..called_number.."@"..server_ip_address;
> session:setVariable("bypass_media", "true"); //described below!!!
> session:execute("bridge",forwarding_session_string);
> 3. PHP script: get_server_ip.php
> <<< ?ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> LOADED AT THE MOMENT SERVER>>>
> die($less_loaded_server_ip);
>
> All Media Servers should provide information about their current load into
> one common database (dedicated server or routing server can be used for
> database).
> If you use scripts you can update this table on starting a call and after
> call finished.
> Or you can create a job that will query FreeSWITCH and will update the
> database every, say, 10 seconds.
>
> Basically that's all (really!)
> I beleive it can be used for most general purpose scenarios without need of
> third-party tools.
> P.S.
> The only problem we had from the beginning was that RTP media traffic was
> passing thru both servers: RS and Media Server.
> But after we added the following command to the routing script, RTP media
> traffic started to pass only thru Media server, making RS almost free.
> session:setVariable("bypass_media", "true");
>
>
> Best regards,
> vIT
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From vetali100 at gmail.com  Tue Mar 16 06:19:26 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:19:26 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
Message-ID: <a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>

Hi Tihomir,
What do you mean by "took from Mera"?
If someone else already implemented this, could you please provide me a
link?

This solution was invented by me personally, after looking into the
following: http://wiki.freeswitch.org/wiki/SBC_Setup
They use Kamailio layer as routing, I decided that we can avoid this and use
OOB things.

Thanks for your reply, it can add more value to the solution.

Regards,
vIT



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

> this is something you took from Mera .. isn't it ? :)
>
>
>
> On Tue, Mar 16, 2010 at 1:04 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi dear FS community,
>>
>> After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109) and (
>> http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
>> solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
>> use MySQL).
>>
>> It works on few of our servers right now without any issues.
>>
>> Feel free to reuse / improve it or provide your comments if you see
>> something wrong with it.
>>
>>
>> We have one routing server (RS) and several Media Servers (MS).
>> FreeSWITCH is installed on each server in default configuration.
>>
>> --> MS_1
>> RS  --> MS_2
>> --> MS_N
>>
>>
>> (Failover can be implemented by adding one more routing server, but this
>> is out of scope of current description).
>>
>
> routing server is your single point of failure, you should make two of them
> in active/standby mode. What about registered endpoints ? ... where do you
> register them?.... you should have a DB replication in order to have the
> "registered context" distributed between both routing servers
>
>
> So, i'd suggest to bring up a LAN_HA and have a floating address between
> your routing servers.... of course the you should be using ODBC in the core
> to have registrations on both servers.
>
>
>
>
>>
>> Routing server routes calls to media servers based on logic that you can
>> easily define.
>> (Generally speaking, Routing server can route part of the calls onto
>> itself, but to simplify let's say that it will route only to Media Servers).
>>
>> you will be loosing calls on the on the MS if it goes down ... maybe there
> is a way to "replace" media resource for that calls and keep them going even
> after the MS went down.
>
>
>
>>
>> Routing server is used to register subscribers (it will have directory
>> files
>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>> Media servers do NOT contain files with users (security issue? probably
>> should be managed on dialplan stage of media servers by comparing ip
>> addresses of requests).
>>
>>
> consider ODBC in the core!
>
>
>>
>> The whole routing can be presented by the following chain:
>>
>> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
>> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>> "sofia/external/" to determined MS IP address-->[Bridge to fail over media
>> server, optionally]-->End
>>
>> Let's say we need to route all international calls based on less loaded
>> server (other logic can be easily implemented in PHP script as well).
>>
>> *1. RS Dialplan: default.xml*
>> <include>
>>         <extension name="route">
>>                 <condition field="destination_number"
>> expression="^(00)(\d+)">
>>                         <action application="lua" data="route.lua $2"/>
>>                 </condition>
>>         </extension>
>> </include>
>>
>>
>> *2. RS Lua script: route.lua*
>> called_number = argv[1]; //without "00"
>> api = freeswitch.API();
>> server_ip_address = api:executeString("curl
>> http://localhost/get_server_ip.php);
>> forwarding_session_string =
>> "sofia/external/"..called_number.."@"..server_ip_address;
>> session:setVariable("bypass_media", "true"); //described below!!!
>> session:execute("bridge",forwarding_session_string);
>>
>>
> do your bridging in the dialplan not from a script... you will get better
> performances!
>
>
>
>
>> *3. PHP script: get_server_ip.php*
>> <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
>> LOADED AT THE MOMENT SERVER>>>
>> die($less_loaded_server_ip);
>>
>>
>> All Media Servers should provide information about their current load into
>> one common database (dedicated server or routing server can be used for
>> database).
>> If you use scripts you can update this table on starting a call and after
>> call finished.
>> Or you can create a job that will query FreeSWITCH and will update the
>> database every, say, 10 seconds.
>>
>>
>> Basically that's all (really!)
>> I beleive it can be used for most general purpose scenarios without need
>> of third-party tools.
>>
>> P.S.
>> The only problem we had from the beginning was that RTP media traffic was
>> passing thru both servers: RS and Media Server.
>> But after we added the following command to the routing script, RTP media
>> traffic started to pass only thru Media server, making RS almost free.
>> session:setVariable("bypass_media", "true");
>>
>>
>>
> anyhow it is a good start ... with a custom module for media server
> "kepalive" and resource migration it could be a great solution.
>
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Tue Mar 16 06:21:58 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:21:58 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
Message-ID: <a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>

Basically this solution gives me exactly what I need at the moment.

If there is a still special reason to use OpenSIPS, maybe we should expect
opinion of some better FS experts then me :-)

vIT

2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>

> Vit:
>
> Nice email, very informative; curious to find out the reasons why you
> decided to go with FS instead of OpenSip. We soon have to make a
> similar decision and would like to find out the rationale behind it.
>
> Thanks,
>
> Alfredo
>
> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
> wrote:
> > Hi dear FS community,
> > After a brief analysis of 2 solutions using OpenSIPS
> > and Kamailio (
> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement
> my
> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
> server
> > (we use MySQL).
> > It works on few of our servers right now without any issues.
> > Feel free to reuse / improve it or provide your comments if you see
> > something wrong with it.
> >
> > We have one routing server (RS) and several Media Servers (MS).
> > FreeSWITCH is installed on each server in default configuration.
> > --> MS_1
> > RS  --> MS_2
> > --> MS_N
> >
> > (Failover can be implemented by adding one more routing server, but this
> is
> > out of scope of current description).
> > Routing server routes calls to media servers based on logic that you can
> > easily define.
> > (Generally speaking, Routing server can route part of the calls onto
> itself,
> > but to simplify let's say that it will route only to Media Servers).
> >
> > Routing server is used to register subscribers (it will have directory
> files
> > ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> > Media servers do NOT contain files with users (security issue? probably
> > should be managed on dialplan stage of media servers by comparing ip
> > addresses of requests).
> >
> > The whole routing can be presented by the following chain:
> > Call arrives into Default context-->RS default dialplan-->Lua
> script-->PHP
> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
> media
> > server, optionally]-->End
> > Let's say we need to route all international calls based on less loaded
> > server (other logic can be easily implemented in PHP script as well).
> > 1. RS Dialplan: default.xml
> > <include>
> >         <extension name="route">
> >                 <condition field="destination_number"
> > expression="^(00)(\d+)">
> >                         <action application="lua" data="route.lua $2"/>
> >                 </condition>
> >         </extension>
> > </include>
> >
> > 2. RS Lua script: route.lua
> > called_number = argv[1]; //without "00"
> > api = freeswitch.API();
> > server_ip_address = api:executeString("curl
> > http://localhost/get_server_ip.php);
> > forwarding_session_string =
> > "sofia/external/"..called_number.."@"..server_ip_address;
> > session:setVariable("bypass_media", "true"); //described below!!!
> > session:execute("bridge",forwarding_session_string);
> > 3. PHP script: get_server_ip.php
> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> > LOADED AT THE MOMENT SERVER>>>
> > die($less_loaded_server_ip);
> >
> > All Media Servers should provide information about their current load
> into
> > one common database (dedicated server or routing server can be used for
> > database).
> > If you use scripts you can update this table on starting a call and after
> > call finished.
> > Or you can create a job that will query FreeSWITCH and will update the
> > database every, say, 10 seconds.
> >
> > Basically that's all (really!)
> > I beleive it can be used for most general purpose scenarios without need
> of
> > third-party tools.
> > P.S.
> > The only problem we had from the beginning was that RTP media traffic was
> > passing thru both servers: RS and Media Server.
> > But after we added the following command to the routing script, RTP media
> > traffic started to pass only thru Media server, making RS almost free.
> > session:setVariable("bypass_media", "true");
> >
> >
> > Best regards,
> > vIT
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vetali100 at gmail.com  Tue Mar 16 06:31:13 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:31:13 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
Message-ID: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>

Tihomir,
           Could you please add more details to the following?

           "So, i'd suggest to bring up a LAN_HA and have a floating address
between your routing servers.... of course the you should be using ODBC in
the core to have registrations on both servers."

Do you have any links that describe how to implement this kind of fail over?

I thought about configuring 2 DNS servers on 2 Routing servers, so if one
will go down, traffic will be addressed to another, but at the moment I
don't have enough knowledge how to implement this.

Thanks,
vIT



2010/3/16 Vitalii Colosov <vetali100 at gmail.com>

> Basically this solution gives me exactly what I need at the moment.
>
> If there is a still special reason to use OpenSIPS, maybe we should expect
> opinion of some better FS experts then me :-)
>
> vIT
>
> 2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>
>
> Vit:
>>
>> Nice email, very informative; curious to find out the reasons why you
>> decided to go with FS instead of OpenSip. We soon have to make a
>> similar decision and would like to find out the rationale behind it.
>>
>> Thanks,
>>
>> Alfredo
>>
>> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
>> wrote:
>> > Hi dear FS community,
>> > After a brief analysis of 2 solutions using OpenSIPS
>> > and Kamailio (
>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
>> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to
>> implement my
>> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
>> server
>> > (we use MySQL).
>> > It works on few of our servers right now without any issues.
>> > Feel free to reuse / improve it or provide your comments if you see
>> > something wrong with it.
>> >
>> > We have one routing server (RS) and several Media Servers (MS).
>> > FreeSWITCH is installed on each server in default configuration.
>> > --> MS_1
>> > RS  --> MS_2
>> > --> MS_N
>> >
>> > (Failover can be implemented by adding one more routing server, but this
>> is
>> > out of scope of current description).
>> > Routing server routes calls to media servers based on logic that you can
>> > easily define.
>> > (Generally speaking, Routing server can route part of the calls onto
>> itself,
>> > but to simplify let's say that it will route only to Media Servers).
>> >
>> > Routing server is used to register subscribers (it will have directory
>> files
>> >
>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>> > Media servers do NOT contain files with users (security issue? probably
>> > should be managed on dialplan stage of media servers by comparing ip
>> > addresses of requests).
>> >
>> > The whole routing can be presented by the following chain:
>> > Call arrives into Default context-->RS default dialplan-->Lua
>> script-->PHP
>> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
>> media
>> > server, optionally]-->End
>> > Let's say we need to route all international calls based on less loaded
>> > server (other logic can be easily implemented in PHP script as well).
>> > 1. RS Dialplan: default.xml
>> > <include>
>> >         <extension name="route">
>> >                 <condition field="destination_number"
>> > expression="^(00)(\d+)">
>> >                         <action application="lua" data="route.lua $2"/>
>> >                 </condition>
>> >         </extension>
>> > </include>
>> >
>> > 2. RS Lua script: route.lua
>> > called_number = argv[1]; //without "00"
>> > api = freeswitch.API();
>> > server_ip_address = api:executeString("curl
>> > http://localhost/get_server_ip.php);
>> > forwarding_session_string =
>> > "sofia/external/"..called_number.."@"..server_ip_address;
>> > session:setVariable("bypass_media", "true"); //described below!!!
>> > session:execute("bridge",forwarding_session_string);
>> > 3. PHP script: get_server_ip.php
>> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE
>> LESS
>> > LOADED AT THE MOMENT SERVER>>>
>> > die($less_loaded_server_ip);
>> >
>> > All Media Servers should provide information about their current load
>> into
>> > one common database (dedicated server or routing server can be used for
>> > database).
>> > If you use scripts you can update this table on starting a call and
>> after
>> > call finished.
>> > Or you can create a job that will query FreeSWITCH and will update the
>> > database every, say, 10 seconds.
>> >
>> > Basically that's all (really!)
>> > I beleive it can be used for most general purpose scenarios without need
>> of
>> > third-party tools.
>> > P.S.
>> > The only problem we had from the beginning was that RTP media traffic
>> was
>> > passing thru both servers: RS and Media Server.
>> > But after we added the following command to the routing script, RTP
>> media
>> > traffic started to pass only thru Media server, making RS almost free.
>> > session:setVariable("bypass_media", "true");
>> >
>> >
>> > Best regards,
>> > vIT
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mustafa.pk at gmail.com  Tue Mar 16 06:41:05 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Tue, 16 Mar 2010 18:41:05 +0500
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
Message-ID: <8213d6071003160641k4d10c0dcs47c1a94dc05fa080@mail.gmail.com>

On Tue, Mar 16, 2010 at 6:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Tihomir,
>            Could you please add more details to the following?
>
>            "So, i'd suggest to bring up a LAN_HA and have a floating
> address between your routing servers.... of course the you should be using
> ODBC in the core to have registrations on both servers."
>

no problem, you can update your knowledge base for free, check
http://www.linux-ha.org/wiki/Main_Page

>
> Do you have any links that describe how to implement this kind of fail
> over?
>
> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
> will go down, traffic will be addressed to another, but at the moment I
> don't have enough knowledge how to implement this.
>
> Thanks,
> vIT
>
>
>
> 2010/3/16 Vitalii Colosov <vetali100 at gmail.com>
>
> Basically this solution gives me exactly what I need at the moment.
>>
>> If there is a still special reason to use OpenSIPS, maybe we should expect
>> opinion of some better FS experts then me :-)
>>
>> vIT
>>
>> 2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>
>>
>> Vit:
>>>
>>> Nice email, very informative; curious to find out the reasons why you
>>> decided to go with FS instead of OpenSip. We soon have to make a
>>> similar decision and would like to find out the rationale behind it.
>>>
>>> Thanks,
>>>
>>> Alfredo
>>>
>>> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
>>> wrote:
>>> > Hi dear FS community,
>>> > After a brief analysis of 2 solutions using OpenSIPS
>>> > and Kamailio (
>>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
>>> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to
>>> implement my
>>> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
>>> server
>>> > (we use MySQL).
>>> > It works on few of our servers right now without any issues.
>>> > Feel free to reuse / improve it or provide your comments if you see
>>> > something wrong with it.
>>> >
>>> > We have one routing server (RS) and several Media Servers (MS).
>>> > FreeSWITCH is installed on each server in default configuration.
>>> > --> MS_1
>>> > RS  --> MS_2
>>> > --> MS_N
>>> >
>>> > (Failover can be implemented by adding one more routing server, but
>>> this is
>>> > out of scope of current description).
>>> > Routing server routes calls to media servers based on logic that you
>>> can
>>> > easily define.
>>> > (Generally speaking, Routing server can route part of the calls onto
>>> itself,
>>> > but to simplify let's say that it will route only to Media Servers).
>>> >
>>> > Routing server is used to register subscribers (it will have directory
>>> files
>>> >
>>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>>> > Media servers do NOT contain files with users (security issue? probably
>>> > should be managed on dialplan stage of media servers by comparing ip
>>> > addresses of requests).
>>> >
>>> > The whole routing can be presented by the following chain:
>>> > Call arrives into Default context-->RS default dialplan-->Lua
>>> script-->PHP
>>> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>>> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
>>> media
>>> > server, optionally]-->End
>>> > Let's say we need to route all international calls based on less loaded
>>> > server (other logic can be easily implemented in PHP script as well).
>>> > 1. RS Dialplan: default.xml
>>> > <include>
>>> >         <extension name="route">
>>> >                 <condition field="destination_number"
>>> > expression="^(00)(\d+)">
>>> >                         <action application="lua" data="route.lua $2"/>
>>> >                 </condition>
>>> >         </extension>
>>> > </include>
>>> >
>>> > 2. RS Lua script: route.lua
>>> > called_number = argv[1]; //without "00"
>>> > api = freeswitch.API();
>>> > server_ip_address = api:executeString("curl
>>> > http://localhost/get_server_ip.php);
>>> > forwarding_session_string =
>>> > "sofia/external/"..called_number.."@"..server_ip_address;
>>> > session:setVariable("bypass_media", "true"); //described below!!!
>>> > session:execute("bridge",forwarding_session_string);
>>> > 3. PHP script: get_server_ip.php
>>> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE
>>> LESS
>>> > LOADED AT THE MOMENT SERVER>>>
>>> > die($less_loaded_server_ip);
>>> >
>>> > All Media Servers should provide information about their current load
>>> into
>>> > one common database (dedicated server or routing server can be used for
>>> > database).
>>> > If you use scripts you can update this table on starting a call and
>>> after
>>> > call finished.
>>> > Or you can create a job that will query FreeSWITCH and will update the
>>> > database every, say, 10 seconds.
>>> >
>>> > Basically that's all (really!)
>>> > I beleive it can be used for most general purpose scenarios without
>>> need of
>>> > third-party tools.
>>> > P.S.
>>> > The only problem we had from the beginning was that RTP media traffic
>>> was
>>> > passing thru both servers: RS and Media Server.
>>> > But after we added the following command to the routing script, RTP
>>> media
>>> > traffic started to pass only thru Media server, making RS almost free.
>>> > session:setVariable("bypass_media", "true");
>>> >
>>> >
>>> > Best regards,
>>> > vIT
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From anthony.minessale at gmail.com  Tue Mar 16 06:45:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 08:45:29 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
Message-ID: <191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>

Are you bridging to sip or some ivr and making dingaling wait until the call
answers?

gtalk is sort of missing key telephony features like the concept of pdd and
indication progress etc.

Try answering the call in your dialplan before you bridge.

I think the gtalk client is getting impatient because you did not answer
soon enough with an ip choice
so it starts sending all these extra ones at you and gave up on the one we
chose.


On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > FS would prefer you run your kernel at 1000hz you are probably on a VM if
> > its 10000hz
> >
>
> It's a  physical machine. And I'm also running 20 skype clients. as
> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> 250 HZ. weird.
>
>
> here is the dingaling log with dl_debug on
>
> http://pastebin.freeswitch.org/12453
> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>
>
> still get this message on client:
>
> Sorry! The voice chat with language_lesson failed because of a network
> problem at 5:39 PM. Please try again.
>  Click here to report this to Google
>
>
>
> Thanks.
>
>
>
>
>
>
>
>
>
> >
> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> Windows client and web site on Mac. They both work on the other two
> >> instances of FreeSWITCH I mentioned in this thread.
> >>
> >> btw, can you give me a hint on the timer warning?
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > you would have to execute
> >> >
> >> > dl_debug on
> >> >
> >> > to get a trace that would tell us anything about that?
> >> >
> >> > Are you using the windows gtalk client or the web site?
> >> > We are following google who have been pushing for the online client
> >> > embedded
> >> > in the gmail website.
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> here is the client:
> >> >>
> >> >> <profile type="client">
> >> >>  <param name="name" value="gmail.com"/>
> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >>  <param name="password" value="password"/>
> >> >>  <param name="server" value="talk.google.com"/>
> >> >>  <param name="dialplan" value="XML"/>
> >> >>  <param name="contex" value="public"/>
> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >>  <param name="rtp-ip" value="auto"/>
> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >>  <param name="auto-login" value="true"/>
> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >>  <param name="sasl" value="plain"/>
> >> >>  <param name="tls" value="true"/>
> >> >>  <param name="exten" value="5000"/>
> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> >> </profile>
> >> >>
> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> > ALL,
> >> >> >
> >> >> > I just upgrade FS to 16992.
> >> >> >
> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
> sounds
> >> >> > ok.
> >> >> >
> >> >> >
> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> >> >> > detected!
> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> >> > experience audio problems.
> >> >> >
> >> >> > uname -a
> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05
> UTC
> >> >> > 2008 i686 GNU/Linux
> >> >> >
> >> >> >
> >> >> > I used a script to check the interupts
> >> >> >
> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >
> >> >> >
> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> answer.
> >> >> > I
> >> >> > originate a call from FS, the console blocks, gtalk client ring and
> >> >> > it
> >> >> > looks ok when I hit answer, but no sound, FS console still blocks.
> a
> >> >> > few seconds later gtalk client shows:
> >> >> >
> >> >> > Sorry! The voice chat with language_lesson failed because of a
> >> >> > network
> >> >> > problem at 5:50 PM. Please try again.
> >> >> >  Click here to report this to Google.
> >> >> >
> >> >> >
> >> >> > here is a log with bgapi:
> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >
> >> >> >
> >> >> > rev 14696 works on the same server same conf. another server
> running
> >> >> > 16958M works at the same time. I have no clue to check, is it
> related
> >> >> > to the timer as in 1)? can some help me to take a look? let me know
> >> >> > if
> >> >> > you need "dl_debug on" logs.
> >> >> >
> >> >> > Thanks.
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar 16 06:53:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 08:53:33 -0500
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
Message-ID: <5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>

Can you collect some logs and traces to show us this?

/b

On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:

> OK - sorry to make such a meal of this.  



From anthony.minessale at gmail.com  Tue Mar 16 06:54:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 08:54:33 -0500
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
Message-ID: <191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg
command itself
is designed to send a 1 way message to the channel, the reply is only to
confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only
thing you can do.

if you don't like async behaviour you could always send a blocking
instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000,
NULL);



On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  I agree, that is a possibility. However, when sending a ?SendMsg? with
> application ?break?, and receive ?+OK? as it?s reply, you kind of expect
> that it was received correctly, and will be handled in the event queue
> properly.
>
>
>
> I have a created a small patch that (probably) will fix this in FS, without
> breaking anything else. I just need to test it first ? if everything seems
> ok I will add a jira for it, with the patch attached.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 17:21
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could also wait for the execute_complete of the file you were playing
>
> On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
> On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
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> !DSPAM:4b9e60d032938314917241!
>
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> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From peter.olsson at visionutveckling.se  Tue Mar 16 07:13:40 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 15:13:40 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>

Anthony, thanks for further information!

My goal is not to wait for the break to be executed, I just want to make sure it will be handled properly - and actually break the channel that is playing. So nothing wrong with the async behaviour itself - I like it the way it is.

I've been digging into the FS code for a couple of hours, and I think the problem occurs when I'm "fast" enough to add the break command, and then a new playback command in the event queue (and they both exist there). Then in the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked, switch_ivr_parse_all_events() will be called. If both events exist in the queue, the CF_BREAK will first be set, then on the next event it will be reset again, and the new playback will begin. But it will never break the original playback.

Since I havn't digging too deep in the code, I'm not 100% sure of this :) I was hoping you could help out with your expertice. My current patch looks something like this - do you think it's possible this causes other problems? By the way - sorry for getting this technical in the users-list, I guess this belongs more to the dev-list...

Index: src/switch_ivr.c
===================================================================
--- src/switch_ivr.c                         (revision 17000)
+++ src/switch_ivr.c                      (working copy)
@@ -660,12 +660,16 @@
 SWITCH_DECLARE(switch_status_t) switch_ivr_parse_all_events(switch_core_session_t *session)
 {
                             int x = 0;
+                           switch_channel_t *channel = switch_core_session_get_channel(session);

-
                             switch_ivr_parse_all_messages(session);

-                            while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS)
+                           while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS) {
                                                          x++;
+                                                        if (switch_channel_test_flag(channel, CF_BREAK)) {
+                                                                                     break;
+                                                        }
+        }

                             if (x) {
                                                          switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);

/Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 16 mars 2010 14:55
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg command itself
is designed to send a 1 way message to the channel, the reply is only to confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only thing you can do.

if you don't like async behaviour you could always send a blocking instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000, NULL);


On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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



--
Anthony Minessale II

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

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

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

_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
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_______________________________________________
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--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
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!DSPAM:4b9f8f4132931389420256!
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From moises.silva at gmail.com  Tue Mar 16 07:32:28 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 16 Mar 2010 10:32:28 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
	<c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
	<10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>
Message-ID: <c4d05cbe1003160732r509f534frafeea0921afb78f7@mail.gmail.com>

I'll contact you off-list to arrange ssh debug session.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Tue, Mar 16, 2010 at 12:31 AM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

>
>
> On Mon, Mar 15, 2010 at 5:28 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Ok, so the call is not even being placed in the ISDN network.
>>
>> Did you manage to get the sangoma_prid logs working? if that's the case
>> then paste them too, it seems sangoma_prid is not seeing the request from
>> FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
>> response from sangoma_prid binary (they both communicate through sctp
>> socket).
>>
>
> This is the log (a result of `# /usr/sbin/sangoma_prid &>
> sangoma_prid.log`)
>
> ================System restart=============
> = Sangoma PRI Protocol Stack Daemon       =
> = Version: 1.58                           =
> = Date: Feb 18 2010                       =
> = Wanpipe Release: wanpipe-3.5.10.3          =
> = Revision:Revision: 15364          =
> ===========================================
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.overload.task
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.100%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.90%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.overload.task
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.100%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.90%-full
> WARN - detected an AFT-A101 board with an unsupported firmware version.
> (detected: 0x34, supported: 0x36). You may encounter unknown issues if you
> use an unsupported firmware version. Please contact support to get the
> proper instructions to get the supported firmware unless you have been
> instructed to use this firmware version.
> ERROR - Assertion failed (in_eNewState == STARTING_TSLINK_STATE)
> file=../../../stack/src/main.cpp line=134
> ERROR - ACE_Task::wait failed with error 22 (Invalid argument) . Task name
> = TelesoftStack
> ERROR - failed to waitUntilDone the TaskManager's thread TelesoftStack
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From anthony.minessale at gmail.com  Tue Mar 16 07:33:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 09:33:40 -0500
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
Message-ID: <191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>

my last email had a code suggestion

On Tue, Mar 16, 2010 at 9:13 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Anthony, thanks for further information!
>
>
>
> My goal is not to wait for the break to be executed, I just want to make
> sure it will be handled properly ? and actually break the channel that is
> playing. So nothing wrong with the async behaviour itself - I like it the
> way it is.
>
>
>
> I?ve been digging into the FS code for a couple of hours, and I think the
> problem occurs when I?m ?fast? enough to add the break command, and then a
> new playback command in the event queue (and they both exist there). Then in
> the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked,
> switch_ivr_parse_all_events() will be called. If both events exist in the
> queue, the CF_BREAK will first be set, then on the next event it will be
> reset again, and the new playback will begin. But it will never break the
> original playback.
>
>
>
> Since I havn?t digging too deep in the code, I?m not 100% sure of this :) I
> was hoping you could help out with your expertice. My current patch looks
> something like this ? do you think it?s possible this causes other problems?
> By the way ? sorry for getting this technical in the users-list, I guess
> this belongs more to the dev-list...
>
>
>
> Index: src/switch_ivr.c
>
> ===================================================================
>
> --- src/switch_ivr.c                         (revision 17000)
>
> +++ src/switch_ivr.c                      (working copy)
>
> @@ -660,12 +660,16 @@
>
>  SWITCH_DECLARE(switch_status_t)
> switch_ivr_parse_all_events(switch_core_session_t *session)
>
>  {
>
>                              int x = 0;
>
> +                           switch_channel_t *channel =
> switch_core_session_get_channel(session);
>
>
>
> -
>
>                              switch_ivr_parse_all_messages(session);
>
>
>
> -                            while (switch_ivr_parse_next_event(session) ==
> SWITCH_STATUS_SUCCESS)
>
> +                           while (switch_ivr_parse_next_event(session) ==
> SWITCH_STATUS_SUCCESS) {
>
>                                                           x++;
>
> +                                                        if
> (switch_channel_test_flag(channel, CF_BREAK)) {
>
> +
> break;
>
> +                                                        }
>
> +        }
>
>
>
>                              if (x) {
>
>
> switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);
>
>
>
> /Peter Olsson
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 16 mars 2010 14:55
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> These events sent via sendmsg are designed to be asynchronous.
> artificial waiting may be adding to the problem.
>
> Step outside of what you are trying to do and consider that the sendmsg
> command itself
> is designed to send a 1 way message to the channel, the reply is only to
> confirm that it was sent not the reply from
> the application you are executing, because executing an app is not the only
> thing you can do.
>
> if you don't like async behaviour you could always send a blocking
> instruction like the read or play_and_get_digits app to collect your info.
>
>
> If anything, it might be ok to put this in the "break" app at the end
>
> switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000,
> NULL);
>
>
>  On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I agree, that is a possibility. However, when sending a ?SendMsg? with
> application ?break?, and receive ?+OK? as it?s reply, you kind of expect
> that it was received correctly, and will be handled in the event queue
> properly.
>
>
>
> I have a created a small patch that (probably) will fix this in FS, without
> breaking anything else. I just need to test it first ? if everything seems
> ok I will add a jira for it, with the patch attached.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 17:21
>
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could also wait for the execute_complete of the file you were playing
>
> On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
> On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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>
>
> _______________________________________________
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>
>
>
> --
> Anthony Minessale II
>
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>
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> pstn:+19193869900
>
>
> _______________________________________________
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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>


-- 
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From peter.olsson at visionutveckling.se  Tue Mar 16 08:00:14 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 16:00:14 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
	<191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577435208@cooper>

Yes, I noticed that - thanks again :)

What exactly would that line do, I dont know how the CF_BROADCAST flag works.. What exacly would it wait for?

Just one more question about my patch - what I'm trying to do is to make sure that CF_BREAK flag is handled in switch_ivr_play_file() (and others calling switch_ivr_parse_all_events() in the loop), even when the event queue holds more than one event (first event "break", and next one "playback", which will reset CF_BREAK flag again). As it works right now CF_BREAK will not be handled in the first playback loop, if the queue holds a break event and then a new playback - is that really the way it should be?

Sorry for all questions, I just want know more, and understand the FS code a bite more...

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 16 mars 2010 15:34
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

my last email had a code suggestion
On Tue, Mar 16, 2010 at 9:13 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Anthony, thanks for further information!

My goal is not to wait for the break to be executed, I just want to make sure it will be handled properly - and actually break the channel that is playing. So nothing wrong with the async behaviour itself - I like it the way it is.

I've been digging into the FS code for a couple of hours, and I think the problem occurs when I'm "fast" enough to add the break command, and then a new playback command in the event queue (and they both exist there). Then in the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked, switch_ivr_parse_all_events() will be called. If both events exist in the queue, the CF_BREAK will first be set, then on the next event it will be reset again, and the new playback will begin. But it will never break the original playback.

Since I havn't digging too deep in the code, I'm not 100% sure of this :) I was hoping you could help out with your expertice. My current patch looks something like this - do you think it's possible this causes other problems? By the way - sorry for getting this technical in the users-list, I guess this belongs more to the dev-list...

Index: src/switch_ivr.c
===================================================================
--- src/switch_ivr.c                         (revision 17000)
+++ src/switch_ivr.c                      (working copy)
@@ -660,12 +660,16 @@
 SWITCH_DECLARE(switch_status_t) switch_ivr_parse_all_events(switch_core_session_t *session)
 {
                             int x = 0;
+                           switch_channel_t *channel = switch_core_session_get_channel(session);

-
                             switch_ivr_parse_all_messages(session);

-                            while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS)
+                           while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS) {
                                                          x++;
+                                                        if (switch_channel_test_flag(channel, CF_BREAK)) {
+                                                                                     break;
+                                                        }
+        }

                             if (x) {
                                                          switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);

/Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 16 mars 2010 14:55

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg command itself
is designed to send a 1 way message to the channel, the reply is only to confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only thing you can do.

if you don't like async behaviour you could always send a blocking instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000, NULL);

On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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From jbrucehopkins at gmail.com  Tue Mar 16 08:05:54 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 15:05:54 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
	<5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
	<d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>
Message-ID: <d03a10e01003160805t36c4f8dbhedaecaf0626df9ae@mail.gmail.com>

... and here are two more Wireshark traces showing the successfull call
scenario, where phone A is configured to use only Sppex (rather than using
Alaw and requiring Freeswitch to transcode).

thanks again
Bruce

On 16 March 2010 14:41, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Sure.  I put the Freeswitch log in pastebin.  here are two Wireshark
> captures in the condition where the call fails- run on two PC's running
> Eyebeam clients.  Phone A initiates the call with only PCMA enabled.  Phone
> B has only Speex enabled.
>
> Thanks for taking the time to look at these.
> Bruce
>
>
> On 16 March 2010 13:53, Brian West <brian at freeswitch.org> wrote:
>
>> Can you collect some logs and traces to show us this?
>>
>> /b
>>
>> On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:
>>
>> > OK - sorry to make such a meal of this.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From ivdreg at gmail.com  Tue Mar 16 08:20:58 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 16 Mar 2010 17:20:58 +0200
Subject: [Freeswitch-users] Mod_limit questions
Message-ID: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>

Hi All,

I have some questions regarding mod_limit.

1. I'm using 16674 in production but tested and with 17001. I try mod_limit
with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash
never tries to write data to SQL. It works (counts sessions) on log screen
but no data in MySQL/Sqlite. It seems that data is cached but cache never
flushed to DB. Is anyone has this problem. My limits are:

<action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
${destination_number} 5 !INCOMING_CALL_BARRED" />
<action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />

and they are applied on inbound call profile in same extension.


2. I need to implement rule maximum of 50 calls per day/24h from the same
CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
example "Limit calls to 5 calls every 10 minutes" is about concurrent call
but not calls processed as count. Any idea ?

Thanks for any help
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From tculjaga at gmail.com  Tue Mar 16 08:38:05 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 16:38:05 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
Message-ID: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Tihomir,
>            Could you please add more details to the following?
>
>            "So, i'd suggest to bring up a LAN_HA and have a floating
> address between your routing servers.... of course the you should be using
> ODBC in the core to have registrations on both servers."
>
> Do you have any links that describe how to implement this kind of fail
> over?
>
>
This is the topic for you:
http://www.ultramonkey.org/3/topologies/
and this is what i already have implemented:
http://www.ultramonkey.org/3/topologies/ha-eg.html




> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
> will go down, traffic will be addressed to another, but at the moment I
> don't have enough knowledge how to implement this.
>
>
don't like the idea as you are introducing one more point of failure :)


> Thanks,
> vIT
>
>
>
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From tculjaga at gmail.com  Tue Mar 16 08:38:47 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 16:38:47 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
Message-ID: <65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:19 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi Tihomir,
> What do you mean by "took from Mera"?
> If someone else already implemented this, could you please provide me a
> link?
>
>
>
>
read the file attached :)


T.
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From anthony.minessale at gmail.com  Tue Mar 16 08:48:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 10:48:20 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
Message-ID: <191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>

depending on the phone FreeSWITCH will update the display any time a call is
bridged.

it works on polycom,snom and aastra and I am not sure if there are more.


On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:

> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
>
> When a call is received, the extension really needs to display information
> about the origin and the intended recipient of the call. Normally for an
> incoming external call, the former is supplied by the 'incoming CLI' (often
> linked to locally stored name) and the latter probably by associating a name
> with each incoming line, so in this way the recipient can see at a glance
> who called whom (or what).
>
> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
>
> So, how does FS handle all the above?
> What does it pass to the recipient, for both external and internal calls?
> Is this part of FS configuration? IOW, do you have programmatic access to
> such information and can you specify exactly what is sent to an extension?
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From srinivas.ksvreddy at gmail.com  Tue Mar 16 08:49:03 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 16 Mar 2010 21:19:03 +0530
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>

we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to
use new versions.

srinivas

On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
> like just farwading the call.
> <action application="bridge" data=
> sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
> / <data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>>
> i do not using any gateway here, just farwarding call to router, is this
> correct way of doing? any idea
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From vetali100 at gmail.com  Tue Mar 16 08:51:19 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 17:51:19 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
Message-ID: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>

Using it you still have one point of failure - only one router and it's on
one internet line...

What if one of these things goes down for a period? :-)

I would prefer to have 2 separate servers on 2 different internet lines
(better different providers even).

Thanks,
vIT


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

>
>
> On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Tihomir,
>>            Could you please add more details to the following?
>>
>>            "So, i'd suggest to bring up a LAN_HA and have a floating
>> address between your routing servers.... of course the you should be using
>> ODBC in the core to have registrations on both servers."
>>
>> Do you have any links that describe how to implement this kind of fail
>> over?
>>
>>
> This is the topic for you:
> http://www.ultramonkey.org/3/topologies/
> and this is what i already have implemented:
> http://www.ultramonkey.org/3/topologies/ha-eg.html
>
>
>
>
>> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
>> will go down, traffic will be addressed to another, but at the moment I
>> don't have enough knowledge how to implement this.
>>
>>
> don't like the idea as you are introducing one more point of failure :)
>
>
>> Thanks,
>> vIT
>>
>>
>>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Tue Mar 16 08:51:19 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 17:51:19 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
Message-ID: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>

Using it you still have one point of failure - only one router and it's on
one internet line...

What if one of these things goes down for a period? :-)

I would prefer to have 2 separate servers on 2 different internet lines
(better different providers even).

Thanks,
vIT


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

>
>
> On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Tihomir,
>>            Could you please add more details to the following?
>>
>>            "So, i'd suggest to bring up a LAN_HA and have a floating
>> address between your routing servers.... of course the you should be using
>> ODBC in the core to have registrations on both servers."
>>
>> Do you have any links that describe how to implement this kind of fail
>> over?
>>
>>
> This is the topic for you:
> http://www.ultramonkey.org/3/topologies/
> and this is what i already have implemented:
> http://www.ultramonkey.org/3/topologies/ha-eg.html
>
>
>
>
>> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
>> will go down, traffic will be addressed to another, but at the moment I
>> don't have enough knowledge how to implement this.
>>
>>
> don't like the idea as you are introducing one more point of failure :)
>
>
>> Thanks,
>> vIT
>>
>>
>>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 16 09:00:05 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:00:05 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <DC993C45F9704BE585B4D3A344911A31@MOBILEE1705>

so share your customized version to the community and maybe
we ll be glad to help you.... this is the principle of open source
  ----- Original Message ----- 
  From: srinivasula reddy 
  To: freeswitch-users 
  Sent: Tuesday, March 16, 2010 11:49 AM
  Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


  we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.

  srinivas


  On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:

    Hi,

    i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. 
    <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>
    i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea

    -- 
    Srinivasula Reddy K




  -- 
  Srinivasula Reddy K



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From msc at freeswitch.org  Tue Mar 16 09:01:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 09:01:36 -0700
Subject: [Freeswitch-users] Set/Export channel variable [SOLVED]
In-Reply-To: <4295808702047090347@unknownmsgid>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
	<8259813661645760166@unknownmsgid>
	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
	<4295808702047090347@unknownmsgid>
Message-ID: <87f2f3b91003160901x282ab10x6891be46462401d2@mail.gmail.com>

On Mon, Mar 15, 2010 at 11:21 PM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>wrote:

>  So it seems that the FS console by default only displays the ?a? leg vars
> which is why I couldn?t see it (and it makes sense too since gateway is on
> the b leg side). I have the ESL script now display both the inbound/outbound
> after which I can see the sip_gateway_name var.
>
Good catch!
-MC
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From msc at freeswitch.org  Tue Mar 16 08:59:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 08:59:58 -0700
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>

On Tue, Mar 16, 2010 at 8:49 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to
> use new versions.
>
>
Bummer. Unfortunately you are in a precarious situation because no one here
has used 1.0.2 for more than a year and the devs are not exactly fond of
that particular revision. The good news is that FS is very modular, so
unless you've been messing around with the core or mod_sofia you might be
able to update to 1.0.5 and reapply your customization.
-MC
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From brian at freeswitch.org  Tue Mar 16 08:59:53 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 10:59:53 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <C4A9FC2C-A475-4557-BA52-13A734BC75A3@freeswitch.org>

At any time did you happen to think ... "I should join the project and help out."?

/b

On Mar 16, 2010, at 10:49 AM, srinivasula reddy wrote:

> we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.
>  
> srinivas



From msc at freeswitch.org  Tue Mar 16 09:04:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 09:04:14 -0700
Subject: [Freeswitch-users] answering call in irly_media mode
In-Reply-To: <4B9F3A97.4030609@gmail.com>
References: <4B9F3A97.4030609@gmail.com>
Message-ID: <87f2f3b91003160904q59aa4cd3u884b85e3d3ec312f@mail.gmail.com>

On Tue, Mar 16, 2010 at 1:00 AM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> hi all
> i want to setup a custom ringback tone for inbound (not outbound calls)
> is in irly_media mode but i still heare the normal US ringback.
> where is my dialplan extension:
> http://pastebin.freeswitch.org/12452
> thanks
>
> Check out the ringback variable:
http://wiki.freeswitch.org/wiki/Channel_Variables#ringback

-MC
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From rupa at rupa.com  Tue Mar 16 08:58:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 10:58:21 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
Message-ID: <b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>

1) limit_hash use an in-memory hash for it's storage. There is nor storing
to a database when using limit_hash.  If you want to use the database, then
use plain limit.

2) If you want a max calls in a 24hrs period use you rate with limit_hash.
 This should be about a true max per period, not concurrent.

limit_hash realm id 50/86400

should do what you want -- from memory.  double check the wiki/help to
verify syntax.

On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi All,
>
> I have some questions regarding mod_limit.
>
> 1. I'm using 16674 in production but tested and with 17001. I try mod_limit
> with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash
> never tries to write data to SQL. It works (counts sessions) on log screen
> but no data in MySQL/Sqlite. It seems that data is cached but cache never
> flushed to DB. Is anyone has this problem. My limits are:
>
> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
> ${destination_number} 5 !INCOMING_CALL_BARRED" />
> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>
> and they are applied on inbound call profile in same extension.
>
>
> 2. I need to implement rule maximum of 50 calls per day/24h from the same
> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
> but not calls processed as count. Any idea ?
>
> Thanks for any help
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Tue Mar 16 09:09:25 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 11:09:25 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
	<87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>
Message-ID: <B6737CFC-B597-4433-B99B-F8F12DD6F3F5@freeswitch.org>

1.0.2 was used for exactly two weeks before 1.0.3 came out.. 1.0.2 has a boat load of bugs.

/b

On Mar 16, 2010, at 10:59 AM, Michael Collins wrote:

> Bummer. Unfortunately you are in a precarious situation because no one here has used 1.0.2 for more than a year and the devs are not exactly fond of that particular revision. The good news is that FS is very modular, so unless you've been messing around with the core or mod_sofia you might be able to update to 1.0.5 and reapply your customization. 
> -MC



From ken at ukgb.net  Tue Mar 16 09:23:51 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 16:23:51 +0000
Subject: [Freeswitch-users] call info display
In-Reply-To: <191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
Message-ID: <3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>

Thanks, but could I ask for more specifics? Exactly what info is sent and is it sent as part of the SIP info, i.e. so any SIP device ought to display it correctly? Are there any docs I can read that will explain in detail about the information that is sent (i.e. call info, not SIP protocol stuff) and how to configureFS to send exactly what is required?


On 16 Mar 2010, at 15:48, Anthony Minessale wrote:

> depending on the phone FreeSWITCH will update the display any time a call is bridged.
> 
> it works on polycom,snom and aastra and I am not sure if there are more.
> 
> 
> On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
>> Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.
>> 
>> When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).
>> 
>> Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?
>> 
>> So, how does FS handle all the above?
>> What does it pass to the recipient, for both external and internal calls?
>> Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?
> 


Ken G i l l e t t

_/_/_/_/_/_/_/_/





From vetali100 at gmail.com  Tue Mar 16 09:28:52 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 18:28:52 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
	<65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
Message-ID: <a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>

I have read the document and except that it describes one signalling server
and three media servers with routing to less busiest server it has nothing
else related to solution I wrote for FreeSWITCH.
Basically topology and high level part will be repeated in any solutions,
because this is obvious for everyone.
The same topology as described in this document and by me will be created
using any routing server (OpenSIPS or whatever) and will come to the mind of
most people that need to implement load balancing.


Regards,
vIT



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

>
>
> On Tue, Mar 16, 2010 at 2:19 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi Tihomir,
>> What do you mean by "took from Mera"?
>> If someone else already implemented this, could you please provide me a
>> link?
>>
>>
>>
>>
> read the file attached :)
>
>
> T.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 16 09:34:36 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:34:36 -0400
Subject: [Freeswitch-users] modules update
Message-ID: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>

Hi there,

do I need to "make install" separatly modules (like distributor)
everytime I upgrade from trunk (make current) ?

Thanks

Franck
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From testeador01 at gmail.com  Tue Mar 16 09:39:34 2010
From: testeador01 at gmail.com (Milena)
Date: Tue, 16 Mar 2010 11:39:34 -0500
Subject: [Freeswitch-users] modules update
Message-ID: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>

Hello,

if you edit modules.conf (in the source folder) and uncomment every module
that you want to build, you will not need to make install every module, make
current is enough.

:)
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From brian at freeswitch.org  Tue Mar 16 09:41:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 11:41:28 -0500
Subject: [Freeswitch-users] modules update
In-Reply-To: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
References: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
Message-ID: <1BBBD2D5-577E-401D-9560-96DA87CDA34A@freeswitch.org>

Make sure modules.conf has the module in it and type "make current" it will rebuild everything and update you to the latest SVN.

/b

On Mar 16, 2010, at 11:34 AM, Madovsky wrote:

> Hi there,
>  
> do I need to "make install" separatly modules (like distributor)
> everytime I upgrade from trunk (make current) ?
>  
> Thanks
>  
> Franck

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From rupa at rupa.com  Tue Mar 16 09:43:14 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 11:43:14 -0500
Subject: [Freeswitch-users] modules update
In-Reply-To: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
References: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
Message-ID: <b39273531003160943r3b89c3fbg91c915649b91ee02@mail.gmail.com>

make current

is enough if your modules.conf has the modules you are interested
uncommented.

On Tue, Mar 16, 2010 at 11:34 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi there,
>
> do I need to "make install" separatly modules (like distributor)
> everytime I upgrade from trunk (make current) ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From infos at madovsky.org  Tue Mar 16 09:48:37 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:48:37 -0400
Subject: [Freeswitch-users] modules update
References: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>
Message-ID: <99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>

Ok I was not unsure since on wiki (for distributor module)
it says "make mod_distributor" and "make mod_distributor-install"
to install it properly.

thanks !
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:39 PM
  Subject: Re: [Freeswitch-users] modules update


  Hello,


  if you edit modules.conf (in the source folder) and uncomment every module that you want to build, you will not need to make install every module, make current is enough.


  :)


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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From tculjaga at gmail.com  Tue Mar 16 09:58:51 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 17:58:51 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
	<a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>
Message-ID: <65d96fc81003160958n274d7a18r4b952dcf451d9202@mail.gmail.com>

On Tue, Mar 16, 2010 at 4:51 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Using it you still have one point of failure - only one router and it's on
> one internet line...
>
> What if one of these things goes down for a period? :-)
>
>
well, thats something out of our reach ... we are taking care about the
application and services that are working on top.... well the network is a
totally different topic and it has to be addressed with care.


> I would prefer to have 2 separate servers on 2 different internet lines
> (better different providers even).
>
>
right ... with BGP/MPLS in between as well ... as i said it is a different
topic :)


> Thanks,
> vIT
>
>
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From tculjaga at gmail.com  Tue Mar 16 10:00:56 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 18:00:56 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
	<65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
	<a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>
Message-ID: <65d96fc81003161000g29703556je24c88cace064e2a@mail.gmail.com>

On Tue, Mar 16, 2010 at 5:28 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> I have read the document and except that it describes one signalling server
> and three media servers with routing to less busiest server it has nothing
> else related to solution I wrote for FreeSWITCH.
> Basically topology and high level part will be repeated in any solutions,
> because this is obvious for everyone.
> The same topology as described in this document and by me will be created
> using any routing server (OpenSIPS or whatever) and will come to the mind of
> most people that need to implement load balancing.
>
>
> Regards,
> vIT
>
>
> it is just the topology that is the same ... RS + MS, after that you are on
your own... that's what i meant.

T.
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From anthony.minessale at gmail.com  Tue Mar 16 10:11:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 12:11:56 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
Message-ID: <191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>

as i said it is phone specific.

The callee id and caleeid number is sent

on polycom and astra via sip UPDATE
on snom and FS->FS via INFO


There is no configuration necessary, its the default behavior.


On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:

> Thanks, but could I ask for more specifics? Exactly what info is sent and
> is it sent as part of the SIP info, i.e. so any SIP device ought to display
> it correctly? Are there any docs I can read that will explain in detail
> about the information that is sent (i.e. call info, not SIP protocol stuff)
> and how to configureFS to send exactly what is required?
>
>
> On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
>
> > depending on the phone FreeSWITCH will update the display any time a call
> is bridged.
> >
> > it works on polycom,snom and aastra and I am not sure if there are more.
> >
> >
> > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> >> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
> >>
> >> When a call is received, the extension really needs to display
> information about the origin and the intended recipient of the call.
> Normally for an incoming external call, the former is supplied by the
> 'incoming CLI' (often linked to locally stored name) and the latter probably
> by associating a name with each incoming line, so in this way the recipient
> can see at a glance who called whom (or what).
> >>
> >> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
> >>
> >> So, how does FS handle all the above?
> >> What does it pass to the recipient, for both external and internal
> calls?
> >> Is this part of FS configuration? IOW, do you have programmatic access
> to such information and can you specify exactly what is sent to an
> extension?
> >
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From william.suffill at gmail.com  Tue Mar 16 11:29:32 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 14:29:32 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003141811.58645.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>

FSComm interfaces with sound devices via portaudio at this point. If
portaudio can't use your devices correctly that would cause it not to work
in any portaudio application including FSComm.

What version of Fedora? I'm sure this is fixable but really depends more on
the underlying system more than code in FSComm.


-- W
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From msc at freeswitch.org  Tue Mar 16 11:32:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 11:32:17 -0700
Subject: [Freeswitch-users] modules update
In-Reply-To: <99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>
References: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>
	<99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>
Message-ID: <87f2f3b91003161132y7f377c82jed099fd4d93aa0d8@mail.gmail.com>

On Tue, Mar 16, 2010 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:

>  Ok I was not unsure since on wiki (for distributor module)
> it says "make mod_distributor" and "make mod_distributor-install"
> to install it properly.
>
> thanks !
>

You can do "make mod_<module-name>-install" for any module. It's handy if
you, for example, just did a make current and realized that you hadn't built
mod_distributor. You can edit modules.conf but you don't have to run through
a whole make current again...

-MC

>  ----- Original Message -----
> *From:* Milena <testeador01 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 12:39 PM
> *Subject:* Re: [Freeswitch-users] modules update
>
> Hello,
>
> if you edit modules.conf (in the source folder) and uncomment every module
> that you want to build, you will not need to make install every module, make
> current is enough.
>
> :)
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at ukgb.net  Tue Mar 16 11:52:34 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 18:52:34 +0000
Subject: [Freeswitch-users] call info display
In-Reply-To: <191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
Message-ID: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>

Previously I had a company with a call centre, in which we used a Meridian Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of DDI numbers. Each client was provided with one of those DDI numbers on which their customers would call us, so it was vital the operator knew which client's customer was calling. Within the configuration of the Option 11 I named each incoming DDI with the respective client's (Company) name and then that name was sent to the extension, along with the incoming CLI, so each operator knew which client's number had been called (and hence how to respond) and also who was calling, either their number, or name if it could be matched in an address directory (although cannot now remember if that was done by the switch itself or the extension phone).

Nothing strange about any of that, but I'm trying to relate those same requirements to the capabilities of FS. My assumption has been that FS would be able to match anything traditional telephone switches could manage, but I now need some confirmation of this.

So, are the callee ID and number the incoming line name and number respectively and it's those that FS sends to the extension for display?

Where does this data come from? Surely naming the incoming lines must be part of the configuration.

Can I name them to whatever I want?

Is the incoming CLI number sent directly to the extension (where it's up to the extension what it then displays)?
Or
Does FS do any matching to a name in a directory first and then send the matched name to the extension?

Sorry to keep asking - I'm sure this is very obvious to those who know, but I'm still trying to learn all about FS and what it can and can't do.


On 16 Mar 2010, at 17:11, Anthony Minessale wrote:

> as i said it is phone specific.
> 
> The callee id and caleeid number is sent 
> 
> on polycom and astra via sip UPDATE
> on snom and FS->FS via INFO
> 
> 
> There is no configuration necessary, its the default behavior.
> 
> 
> On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:
> Thanks, but could I ask for more specifics? Exactly what info is sent and is it sent as part of the SIP info, i.e. so any SIP device ought to display it correctly? Are there any docs I can read that will explain in detail about the information that is sent (i.e. call info, not SIP protocol stuff) and how to configureFS to send exactly what is required?
> 
> 
> On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
> 
> > depending on the phone FreeSWITCH will update the display any time a call is bridged.
> >
> > it works on polycom,snom and aastra and I am not sure if there are more.
> >
> >
> > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> >> Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.
> >>
> >> When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).
> >>
> >> Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?
> >>
> >> So, how does FS handle all the above?
> >> What does it pass to the recipient, for both external and internal calls?
> >> Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?
> >



Ken G i l l e t t

_/_/_/_/_/_/_/_/





From ivdreg at gmail.com  Tue Mar 16 11:58:17 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 16 Mar 2010 20:58:17 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
Message-ID: <b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>

Thanks Rupa,

This was good clarification. I want to use mod_limit via 4 servers but
because traffic is equally balanced between them I will use limit_hash and
calculate how many calls every server will serve.

One more question ? Is it necessary limit to transfer to limit_exceeded or
there is a way to set directly disconnect cause like in limit_hash ?

Cheers

2010/3/16 Rupa Schomaker <rupa at rupa.com>

> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
> to a database when using limit_hash.  If you want to use the database, then
> use plain limit.
>
> 2) If you want a max calls in a 24hrs period use you rate with limit_hash.
>  This should be about a true max per period, not concurrent.
>
> limit_hash realm id 50/86400
>
> should do what you want -- from memory.  double check the wiki/help to
> verify syntax.
>
> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi All,
>>
>> I have some questions regarding mod_limit.
>>
>> 1. I'm using 16674 in production but tested and with 17001. I try
>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>
>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>
>> and they are applied on inbound call profile in same extension.
>>
>>
>> 2. I need to implement rule maximum of 50 calls per day/24h from the same
>> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
>> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
>> but not calls processed as count. Any idea ?
>>
>> Thanks for any help
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Mar 16 12:10:58 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 16 Mar 2010 12:10:58 -0700
Subject: [Freeswitch-users] mod_cdr_csv
Message-ID: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>

What triggers mod_cdr_csv to log a call?  

I'm not sure if what I'm experiencing is due to my mistake or otherwise, but here's the situation.
Incoming call (via PSTN or a VoIP provider) -> Extension 105 and answered
105 transfers to 115, 115 answers, 105 completes transfer
The call progresses and is 115 finally hangs up.

The Master.csv only gets one line added to it - the 105 to 115 call.  Nothing about the incoming caller.

If I knew what triggered the log entry, I could try to track the issue down in my dial plan.

Thanks!
-Troy



From frank at carmickle.com  Tue Mar 16 12:12:13 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 16 Mar 2010 15:12:13 -0400
Subject: [Freeswitch-users] call info display
In-Reply-To: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
	<94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
Message-ID: <20100316191212.GD21534@base.carmickle.com>

Hello

On Tue, Mar 16, Ken Gillett wrote:
> Previously I had a company with a call centre, in which we used a Meridian Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of DDI numbers. Each client was provided with one of those DDI numbers on which their customers would call us, so it was vital the operator knew which client's customer was calling. Within the configuration of the Option 11 I named each incoming DDI with the respective client's (Company) name and then that name was sent to the extension, along with the incoming CLI, so each operator knew which client's number had been called (and hence how to respond) and also who was calling, either their number, or name if it could be matched in an address directory (although cannot now remember if that was done by the switch itself or the extension phone).
> 
> Nothing strange about any of that, but I'm trying to relate those same requirements to the capabilities of FS. My assumption has been that FS would be able to match anything traditional telephone switches could manage, but I now need some confirmation of this.
> 
> So, are the callee ID and number the incoming line name and number respectively and it's those that FS sends to the extension for display?
> 
> Where does this data come from? Surely naming the incoming lines must be part of the configuration.
> 
> Can I name them to whatever I want?
> 
> Is the incoming CLI number sent directly to the extension (where it's up to the extension what it then displays)?
> Or
> Does FS do any matching to a name in a directory first and then send the matched name to the extension?
> 
> Sorry to keep asking - I'm sure this is very obvious to those who know, but I'm still trying to learn all about FS and what it can and can't do.
 
Most anything you want to do can be done.  You really need to set up a lab and get your hands dirty to find out what each scenario requires.  This is the only way you can really learn the ins and outs of FS.

--FC


From anthony.minessale at gmail.com  Tue Mar 16 12:16:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 14:16:46 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
	<94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
Message-ID: <191c3a031003161216m582eba60o58b1ab56999d589@mail.gmail.com>

Its automatically determined based on the info in the SIP traffic from the 2
respective ends of the call.
It's also configurable to be overridden from the dialplan.


On Tue, Mar 16, 2010 at 1:52 PM, Ken Gillett <ken at ukgb.net> wrote:

> Previously I had a company with a call centre, in which we used a Meridian
> Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of
> DDI numbers. Each client was provided with one of those DDI numbers on which
> their customers would call us, so it was vital the operator knew which
> client's customer was calling. Within the configuration of the Option 11 I
> named each incoming DDI with the respective client's (Company) name and then
> that name was sent to the extension, along with the incoming CLI, so each
> operator knew which client's number had been called (and hence how to
> respond) and also who was calling, either their number, or name if it could
> be matched in an address directory (although cannot now remember if that was
> done by the switch itself or the extension phone).
>
> Nothing strange about any of that, but I'm trying to relate those same
> requirements to the capabilities of FS. My assumption has been that FS would
> be able to match anything traditional telephone switches could manage, but I
> now need some confirmation of this.
>
> So, are the callee ID and number the incoming line name and number
> respectively and it's those that FS sends to the extension for display?
>
> Where does this data come from? Surely naming the incoming lines must be
> part of the configuration.
>
> Can I name them to whatever I want?
>
> Is the incoming CLI number sent directly to the extension (where it's up to
> the extension what it then displays)?
> Or
> Does FS do any matching to a name in a directory first and then send the
> matched name to the extension?
>
> Sorry to keep asking - I'm sure this is very obvious to those who know, but
> I'm still trying to learn all about FS and what it can and can't do.
>
>
> On 16 Mar 2010, at 17:11, Anthony Minessale wrote:
>
> > as i said it is phone specific.
> >
> > The callee id and caleeid number is sent
> >
> > on polycom and astra via sip UPDATE
> > on snom and FS->FS via INFO
> >
> >
> > There is no configuration necessary, its the default behavior.
> >
> >
> > On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:
> > Thanks, but could I ask for more specifics? Exactly what info is sent and
> is it sent as part of the SIP info, i.e. so any SIP device ought to display
> it correctly? Are there any docs I can read that will explain in detail
> about the information that is sent (i.e. call info, not SIP protocol stuff)
> and how to configureFS to send exactly what is required?
> >
> >
> > On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
> >
> > > depending on the phone FreeSWITCH will update the display any time a
> call is bridged.
> > >
> > > it works on polycom,snom and aastra and I am not sure if there are
> more.
> > >
> > >
> > > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> > >> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
> > >>
> > >> When a call is received, the extension really needs to display
> information about the origin and the intended recipient of the call.
> Normally for an incoming external call, the former is supplied by the
> 'incoming CLI' (often linked to locally stored name) and the latter probably
> by associating a name with each incoming line, so in this way the recipient
> can see at a glance who called whom (or what).
> > >>
> > >> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
> > >>
> > >> So, how does FS handle all the above?
> > >> What does it pass to the recipient, for both external and internal
> calls?
> > >> Is this part of FS configuration? IOW, do you have programmatic access
> to such information and can you specify exactly what is sent to an
> extension?
> > >
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar 16 12:18:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 15:18:10 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>

I noticed that sendmail makes a segfault when FS is trying to send email
and sendmail has some emails locked in folder queue....

Regards

Franck
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From anthony.minessale at gmail.com  Tue Mar 16 12:29:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 14:29:31 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
Message-ID: <191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>

are you running FS as root or a regular user?
maybe there is a permission or resource limitation problem preventing it
from getting the larger stack space necessary?


On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

>  I noticed that sendmail makes a segfault when FS is trying to send email
> and sendmail has some emails locked in folder queue....
>
> Regards
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From robert.dyck at shaw.ca  Tue Mar 16 12:49:21 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Tue, 16 Mar 2010 12:49:21 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>
Message-ID: <201003161249.22448.robert.dyck@shaw.ca>

F11, KDE4.4.1, pulseaudio

Is portaudio another sound server?

RD

On Tuesday 16 March 2010 11:29:32 am William Suffill wrote:
> FSComm interfaces with sound devices via portaudio at this point. If
> portaudio can't use your devices correctly that would cause it not to work
> in any portaudio application including FSComm.
> 
> What version of Fedora? I'm sure this is fixable but really depends more on
> the underlying system more than code in FSComm.
> 
> 
> -- W


From rupa at rupa.com  Tue Mar 16 12:41:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 14:41:17 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
Message-ID: <b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>

limit and limit_hash should work the same with regards to disconnect cause.
 If you see a difference, let me know.

On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Thanks Rupa,
>
> This was good clarification. I want to use mod_limit via 4 servers but
> because traffic is equally balanced between them I will use limit_hash and
> calculate how many calls every server will serve.
>
> One more question ? Is it necessary limit to transfer to limit_exceeded or
> there is a way to set directly disconnect cause like in limit_hash ?
>
> Cheers
>
> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>
> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
>> to a database when using limit_hash.  If you want to use the database, then
>> use plain limit.
>>
>> 2) If you want a max calls in a 24hrs period use you rate with limit_hash.
>>  This should be about a true max per period, not concurrent.
>>
>> limit_hash realm id 50/86400
>>
>> should do what you want -- from memory.  double check the wiki/help to
>> verify syntax.
>>
>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> I have some questions regarding mod_limit.
>>>
>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>
>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>
>>> and they are applied on inbound call profile in same extension.
>>>
>>>
>>> 2. I need to implement rule maximum of 50 calls per day/24h from the same
>>> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
>>> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
>>> but not calls processed as count. Any idea ?
>>>
>>> Thanks for any help
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From devel at thom.fr.eu.org  Tue Mar 16 12:59:25 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Tue, 16 Mar 2010 20:59:25 +0100
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
	<191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
Message-ID: <006c01cac543$2e079510$8a16bf30$@fr.eu.org>

Are you running debian, I have the same problem on debian x64 (already
mentioned here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.h
tml) and cannot fix it.

I don?t know if it?s a priviledge problem, as I got the same results running
FS with or without ?u.

 

Fran?ois

 

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony
Minessale
Envoy? : mardi 16 mars 2010 20:30
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] voicemail options

 

are you running FS as root or a regular user?
maybe there is a permission or resource limitation problem preventing it
from getting the larger stack space necessary?



On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

I noticed that sendmail makes a segfault when FS is trying to send email

and sendmail has some emails locked in folder queue....

 

Regards

 

Franck


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




-- 
Anthony Minessale II

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

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

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

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From freeswitch.org at todandlorna.com  Tue Mar 16 12:59:46 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Tue, 16 Mar 2010 13:59:46 -0600
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
Message-ID: <4B9FE332.7000809@todandlorna.com>

The way I understand it, it gets triggered by the a leg hanging up, same 
as xml_cdr.  I would imagine the channel changes, and if you were 
comparing the xml_cdr of the same call, you would see the original call 
as well.  You might also have some luck with looking at the b leg cdr, 
if you have that turned on.

That's the extent of my guesswork.

-Tod Hansmann

On 3/16/2010 1:10 PM, Troy Anderson wrote:
> What triggers mod_cdr_csv to log a call?
>
> I'm not sure if what I'm experiencing is due to my mistake or otherwise, but here's the situation.
> Incoming call (via PSTN or a VoIP provider) ->  Extension 105 and answered
> 105 transfers to 115, 115 answers, 105 completes transfer
> The call progresses and is 115 finally hangs up.
>
> The Master.csv only gets one line added to it - the 105 to 115 call.  Nothing about the incoming caller.
>
> If I knew what triggered the log entry, I could try to track the issue down in my dial plan.
>
> Thanks!
> -Troy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    


From infos at madovsky.org  Tue Mar 16 13:18:14 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:18:14 -0400
Subject: [Freeswitch-users] voicemail options
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705><191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
	<006c01cac543$2e079510$8a16bf30$@fr.eu.org>
Message-ID: <2549F4D1FBA54CEA9EAA7B0B370B6AB6@MOBILEE1705>

Hi,

no I'm on Fedora 10 64bits.
I'm going to check the source to find a way

Regards

  ----- Original Message ----- 
  From: devel at thom.fr.eu.org 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 3:59 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

  I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

   

  Fran?ois

   

   

  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
  Envoy? : mardi 16 mars 2010 20:30
  ? : freeswitch-users at lists.freeswitch.org
  Objet : Re: [Freeswitch-users] voicemail options

   

  are you running FS as root or a regular user?
  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

  I noticed that sendmail makes a segfault when FS is trying to send email

  and sendmail has some emails locked in folder queue....

   

  Regards

   

  Franck


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




  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From ahmed.ajmal at gmail.com  Mon Mar 15 12:31:34 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Tue, 16 Mar 2010 00:31:34 +0500
Subject: [Freeswitch-users] Fwd:  Set/Export channel variable
In-Reply-To: <9d22cc171003151228p405bfef3v70662dfc2cdc5914@mail.gmail.com>
References: <9d22cc171003151228p405bfef3v70662dfc2cdc5914@mail.gmail.com>
Message-ID: <9d22cc171003151231k209f47aodb09114431505496@mail.gmail.com>

I am looking at the FS Console and I have a ESL listener socket that is
hooked for HANGUP_COMPLETE event.


- ahmed

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at
breezecom.ae <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>wrote:

>*  Hi
*>*
*>*
*>*
*>* I have now configured the gateways as suggested below but I am not able to
*>* see the sip_gateway_name variable when the call hangup event is complete. I
*>* am on Freeswitch 1.0.5.
*>*
*>*
*>*
*Where are you looking to find the variable?
-MC

From: freeswitch-users-bounces at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

[mailto:freeswitch-users-bounces at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] On
Behalf Of Brian
West
Sent: Wednesday, March 10, 2010 7:08 PM
To: freeswitch-users at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

Subject: Re: [Freeswitch-users] Set/Export channel variable



We already set you a variable called sip_gateway_name when a call is placed
out a gateway.  You're also doing this wrong you should NEVER EVER do

@$ipgwX as that data is in the gateway.  You should simply do
sofia/gateway/gwname/number



/b



On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:





Hi



I am trying to set a channel variable on the bridge application using this:



<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>



The application runs fine, what I now need to do is set/export the 'gw'

channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.



Thanks

Ahmed
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From pavera at gmail.com  Mon Mar 15 23:59:38 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 00:59:38 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
Message-ID: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>

Hello,
Pretty new, but so far very happy with freeswitch!  The setup and config is
really nice, straightforward, and I appreciate the design of the system.
 I've worked with Asterisk pretty regularly for the last 4-5 years, and am
really looking for something more stable, and I think I've found it here!

One issue I've run into, and I'm not sure if its just that my use case is
abnormal, hence the email.

I've worked with PBX systems for a number of years (11 now...), and in all
of the systems I've worked with, call parking was kind of a "roaming"
feature.  By this I mean, you think party X (the intended recipient of the
call) is in the building, but they are not at their desk.  So, you park the
call, and then page, or by some other means attempt to locate party X and
tell them "get to the nearest phone and pick up 5901".  If you can't locate
party X or you can't find them in time, the call should time out back to the
parking party (receptionist) to be handled appropriately (message taken,
sent to voicemail, transferred to party X's boss)

This scenario requires a few things:
1) the receptionist/person who actually receives the call needs to be able
to put the call in a specific place (extension) that doesn't change
2) Party X (the intended recipient) needs to be able to access that specific
place (extension) from any phone inside the company
3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
inadequate, because if the receptionist parks 2 people on 5900, one for
party X and one for party Y, in that order, but party Y gets to a phone
first, they will be connected to party X's caller)
4) The receptionist has to be protected from or provided with sufficient
information to avoid connecting 2 customers with each other (IE the
valet_park in freeswitch is inadequate, because if she parks a caller for X
on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
what can she do?  What if she just forgets and sends that caller to 6001?
 Then the 2 customers are bridged together and thats a big no-no)  Also, I
haven't seen a way to make valet_park timeout?  But maybe I'm wrong
there...

The closest I can get to implementing the above scenario is by using
multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
receptionist phone to "notify" her that someone is already parked in a
certain queue.  Unfortunately, there is no enforcement of this policy by the
system, and if she fat fingers her transfer, she will double up a queue,
creating the problem indicated in 3 above.  The other shortcoming of this
setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
becomes impossible to appropriately handle call park.  If the receptionist
is away from her desk and attempts to answer a call, she can't tell which
parking spots are free at all on a regular 2 line phone.  Also, regular
users are prohibited from using call park as they cannot know which slots
are free (the need for this arises often in certain industries, such as law,
client is speaking with attorney, attorney finishes and says "let me connect
you with my assistant to schedule an appointment and handle x, y, and z for
you.", paralegal is not at desk, atty needs to park call and page his
assistant, maybe they are in the copy room, or the library doing
research...)

The asterisk parking lot seems to solve this problem quite elegantly
(although the stability issues of asterisk undermine it).  It is simple to
use (receptionist and everyone else only needs to remember 1 extension to
park any call), The callers/parking spots are maintained by the system to
prevent double parking, and each caller sits at a specific extension until
they are picked up or time out.

So, the question boils down to how do people using freeswitch view/use call
park?  Is there a "better" roaming feature that I just don't know about?  If
it is just something that hasn't been implemented/thought about this way
yet, any idea what it would take to get a call parking feature that
satisfies these requirements into freeswitch?

Thanks for your time, sorry for the long email, hopefully it explains well
enough what I'm trying to accomplish.

-Tom
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From jbrucehopkins at gmail.com  Tue Mar 16 07:41:48 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 14:41:48 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
	<5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
Message-ID: <d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>

Sure.  I put the Freeswitch log in pastebin.  here are two Wireshark
captures in the condition where the call fails- run on two PC's running
Eyebeam clients.  Phone A initiates the call with only PCMA enabled.  Phone
B has only Speex enabled.

Thanks for taking the time to look at these.
Bruce

On 16 March 2010 13:53, Brian West <brian at freeswitch.org> wrote:

> Can you collect some logs and traces to show us this?
>
> /b
>
> On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:
>
> > OK - sorry to make such a meal of this.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From delorenzodesign at gmail.com  Tue Mar 16 11:06:46 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 14:06:46 -0400
Subject: [Freeswitch-users] Lua - playAndGetDigits
Message-ID: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>

I'm having some trouble getting consistent results with playAndGetDigits,
sometimes the digits are record and other times they're not.  It seemed to
have something to do with how quickly the digits were pressed, but now that
no longer seems to be the case.

I'm trying to have users press a confirmation code, in this case "1111"
although I'd like to make it accept "1" or "11" or "111" or "1111" if
possible.

Can anyone point me in the right direction?


    session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
.. provider .. "/1" .. number_to_call);

    while(session:ready()) do
       session:answer();
       session:setInputCallback("onInput","true");

       --playbackMessage(check_message);
       digits = session:playAndGetDigits(4, 4, 1, 5000, "",
get_digits_message, "", "[1]");
       freeswitch.consoleLog("info", digits .. "\n");
       freeswitch.consoleLog("info", "Did this check out? " ..
(check_message == true and "yes" or "no" .. "\n"));

       -- we got what we're looking for, so we can end this
       if(check_success) then
          freeswitch.consoleLog("info", "A positive response was received
from this check.\n");
          break;
       else
          -- not a positive response
          -- if we reached our max attempts, we're finished move on
          if(attempts == max_attempts) then
            freeswitch.consoleLog("info", "We've reached our maximum
attempts for this number.\n");
            break;
          end
       end
    end
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From pavera at gmail.com  Tue Mar 16 11:06:49 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 12:06:49 -0600
Subject: [Freeswitch-users] Call parking strategy/best practice in
	freeswitch?
Message-ID: <82dae1c01003161106n9f3766cga56623d8729331f6@mail.gmail.com>

Hello,
Pretty new, but so far very happy with freeswitch!  The setup and config is
really nice, straightforward, and I appreciate the design of the system.
 I've worked with Asterisk pretty regularly for the last 4-5 years, and am
really looking for something more stable, and I think I've found it here!

One issue I've run into, and I'm not sure if its just that my use case is
abnormal, hence the email.

I've worked with PBX systems for a number of years (11 now...), and in all
of the systems I've worked with, call parking was kind of a "roaming"
feature.  By this I mean, you think party X (the intended recipient of the
call) is in the building, but they are not at their desk.  So, you park the
call, and then page, or by some other means attempt to locate party X and
tell them "get to the nearest phone and pick up 5901".  If you can't locate
party X or you can't find them in time, the call should time out back to the
parking party (receptionist) to be handled appropriately (message taken,
sent to voicemail, transferred to party X's boss)

This scenario requires a few things:
1) the receptionist/person who actually receives the call needs to be able
to put the call in a specific place (extension) that doesn't change
2) Party X (the intended recipient) needs to be able to access that specific
place (extension) from any phone inside the company
3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
inadequate, because if the receptionist parks 2 people on 5900, one for
party X and one for party Y, in that order, but party Y gets to a phone
first, they will be connected to party X's caller)
4) The receptionist has to be protected from or provided with sufficient
information to avoid connecting 2 customers with each other (IE the
valet_park in freeswitch is inadequate, because if she parks a caller for X
on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
what can she do?  What if she just forgets and sends that caller to 6001?
 Then the 2 customers are bridged together and thats a big no-no)  Also, I
haven't seen a way to make valet_park timeout?  But maybe I'm wrong
there...

The closest I can get to implementing the above scenario is by using
multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
receptionist phone to "notify" her that someone is already parked in a
certain queue.  Unfortunately, there is no enforcement of this policy by the
system, and if she fat fingers her transfer, she will double up a queue,
creating the problem indicated in 3 above.  The other shortcoming of this
setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
becomes impossible to appropriately handle call park.  If the receptionist
is away from her desk and attempts to answer a call, she can't tell which
parking spots are free at all on a regular 2 line phone.  Also, regular
users are prohibited from using call park as they cannot know which slots
are free (the need for this arises often in certain industries, such as law,
client is speaking with attorney, attorney finishes and says "let me connect
you with my assistant to schedule an appointment and handle x, y, and z for
you.", paralegal is not at desk, atty needs to park call and page his
assistant, maybe they are in the copy room, or the library doing
research...)

The asterisk parking lot seems to solve this problem quite elegantly
(although the stability issues of asterisk undermine it).  It is simple to
use (receptionist and everyone else only needs to remember 1 extension to
park any call), The callers/parking spots are maintained by the system to
prevent double parking, and each caller sits at a specific extension until
they are picked up or time out.

So, the question boils down to how do people using freeswitch view/use call
park?  Is there a "better" roaming feature that I just don't know about?  If
it is just something that hasn't been implemented/thought about this way
yet, any idea what it would take to get a call parking feature that
satisfies these requirements into freeswitch?

Thanks for your time, sorry for the long email, hopefully it explains well
enough what I'm trying to accomplish.

-Tom
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From delorenzodesign at gmail.com  Tue Mar 16 12:08:56 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 15:08:56 -0400
Subject: [Freeswitch-users] Lua - playAndGetDigits
In-Reply-To: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>
References: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>
Message-ID: <5c9dcbfb1003161208n12ae669fw77a498d9517152d8@mail.gmail.com>

>
> I'm having some trouble getting consistent results with playAndGetDigits,
> sometimes the digits are record and other times they're not.  It seemed to
> have something to do with how quickly the digits were pressed, but now that
> no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case "1111"
> although I'd like to make it accept "1" or "11" or "111" or "1111" if
> possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
> .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
> get_digits_message, "", "[1]");
>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was received
> from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
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From delorenzodesign at gmail.com  Tue Mar 16 13:21:13 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 16:21:13 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
Message-ID: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>

I'm having some trouble getting consistent results with playAndGetDigits,
sometimes the digits are record and other times they're not.  It seemed to
have something to do with how quickly the digits were pressed, but now that
no longer seems to be the case.

I'm trying to have users press a confirmation code, in this case "1111"
although I'd like to make it accept "1" or "11" or "111" or "1111" if
possible.

Can anyone point me in the right direction?


    session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
.. provider .. "/1" .. number_to_call);

    while(session:ready()) do
       session:answer();
       session:setInputCallback("onInput","true");

       --playbackMessage(check_message);
       digits = session:playAndGetDigits(4, 4, 1, 5000, "",
get_digits_message, "", "[1]");
       freeswitch.consoleLog("info", digits .. "\n");
       freeswitch.consoleLog("info", "Did this check out? " ..
(check_message == true and "yes" or "no" .. "\n"));

       -- we got what we're looking for, so we can end this
       if(check_success) then
          freeswitch.consoleLog("info", "A positive response was received
from this check.\n");
          break;
       else
          -- not a positive response
          -- if we reached our max attempts, we're finished move on
          if(attempts == max_attempts) then
            freeswitch.consoleLog("info", "We've reached our maximum
attempts for this number.\n");
            break;
          end
       end
    end
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From brian at freeswitch.org  Tue Mar 16 13:26:03 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 15:26:03 -0500
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <4B9FE332.7000809@todandlorna.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
Message-ID: <5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>

By default its only logging A legs.  See config file <param name="legs" value="a"/> and change to <param name="legs" value="ab"/>

/b



On Mar 16, 2010, at 2:59 PM, Tod Hansmann wrote:

> The way I understand it, it gets triggered by the a leg hanging up, same 
> as xml_cdr.  I would imagine the channel changes, and if you were 
> comparing the xml_cdr of the same call, you would see the original call 
> as well.  You might also have some luck with looking at the b leg cdr, 
> if you have that turned on.
> 
> That's the extent of my guesswork.
> 
> -Tod Hansmann



From william.suffill at gmail.com  Tue Mar 16 13:27:10 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 16:27:10 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003161249.22448.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com> 
	<201003161249.22448.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>

PortAudio is a free, cross platform,
open-source<http://www.portaudio.com/license.html>,
audio I/O library.  It lets you write simple audio programs in 'C' that will
compile and run on many platforms
<http://www.portaudio.com/status.html>including
*Windows, Macintosh (8,9,X), Unix (OSS), SGI, and BeOS*. PortAudio is
intended to promote the exchange of audio synthesis software between
developers on different platforms.

PortAudio provides a very simple API for recording and/or playing sound
using a simple callback function.  Example programs are included that
synthesize sine waves and pink noise, perform fuzz distortion on a guitar,
list available audio devices, etc.
http://www.portaudio.com/

http://wiki.freeswitch.org/wiki/Mod_portaudio


Technically it is possible to get portaudio based applications to work with
pulseaudio but it seems to be an area of some difficulties now that
pulseaudio is used by default by newer distros.

I've had this same issue on my system w/ pulseaudio since it wants to
control all the sound devices itself. Killing pulseaudio should release the
devices and partaudio should detect them in that case. Not a long term fix I
know but still yet to spend enough time to discover the best alternative.

https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-January/003037.html
seems to indicate that it has been  an area of confusion for some time.

-- W
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From infos at madovsky.org  Tue Mar 16 13:35:15 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:35:15 -0400
Subject: [Freeswitch-users] voicemail options
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705><191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
	<006c01cac543$2e079510$8a16bf30$@fr.eu.org>
Message-ID: <D4EC8B7224F340BEAA2D439BC95DF5A3@MOBILEE1705>

ok, I tried 50 times to leave a message,
it worked only one time.
It seems that sendmail needs to be free to send the email....

  ----- Original Message ----- 
  From: devel at thom.fr.eu.org 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 3:59 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

  I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

   

  Fran?ois

   

   

  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
  Envoy? : mardi 16 mars 2010 20:30
  ? : freeswitch-users at lists.freeswitch.org
  Objet : Re: [Freeswitch-users] voicemail options

   

  are you running FS as root or a regular user?
  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

  I noticed that sendmail makes a segfault when FS is trying to send email

  and sendmail has some emails locked in folder queue....

   

  Regards

   

  Franck


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




  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From infos at madovsky.org  Tue Mar 16 13:42:19 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:42:19 -0400
Subject: [Freeswitch-users] ivr call and secret keycode
Message-ID: <B6EAD9F6B34D4B5095009C8C2A1ECF62@MOBILEE1705>

Hi all,

is anyone already worked on an ivr that make a call so
the callee needs to enter a keycode to validate a form filled before on a website for example ?

Thanks

Franck
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From troy at tlainvestments.com  Tue Mar 16 13:59:58 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 16 Mar 2010 13:59:58 -0700
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
Message-ID: <86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>

Thanks, guys!  Setting the legs param to "ab" brought in lots more cdr data, including the LOSE_RACE legs.  I'm still a bit confused, though - can you explain what is considered a-leg vs. b-leg in this context?  I would have thought the call from outside into the system would constitute an a-leg.

Thanks!
Troy

On Mar 16, 2010, at 1:26 PM, Brian West wrote:

> By default its only logging A legs.  See config file <param name="legs" value="a"/> and change to <param name="legs" value="ab"/>
> 
> /b
> 
> 
> 
> On Mar 16, 2010, at 2:59 PM, Tod Hansmann wrote:
> 
>> The way I understand it, it gets triggered by the a leg hanging up, same 
>> as xml_cdr.  I would imagine the channel changes, and if you were 
>> comparing the xml_cdr of the same call, you would see the original call 
>> as well.  You might also have some luck with looking at the b leg cdr, 
>> if you have that turned on.
>> 
>> That's the extent of my guesswork.
>> 
>> -Tod Hansmann
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Tue Mar 16 14:00:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:00:12 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
Message-ID: <87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>

On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I'm having some trouble getting consistent results with playAndGetDigits,
> sometimes the digits are record and other times they're not.  It seemed to
> have something to do with how quickly the digits were pressed, but now that
> no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case "1111"
> although I'd like to make it accept "1" or "11" or "111" or "1111" if
> possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
> .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
> get_digits_message, "", "[1]");
>

You are specifying a minimum of four and a maximum of four digits. Also,
your regex will match any string that has a digit 1 in it, which may or may
not be what you are looking for. Lastly, you haven't specified a terminator
key (like "#") but you are giving the user only one try and waiting 5000ms
for the entry. Try something like this:

digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
"", "\\d+");

Let us know if that works. Oh, one other thing: you might want to add an
invalid message because playAndGetDigits will handle invalid input for you.
-MC

P.S. - This topic is covered in greater detail in Chapter 7 of the upcoming
FreeSWITCH book. ;)



>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was received
> from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
>
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From robert.dyck at shaw.ca  Tue Mar 16 14:11:40 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Tue, 16 Mar 2010 14:11:40 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca>
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
Message-ID: <201003161411.43845.robert.dyck@shaw.ca>

Killing pulse audio didn't help. Is it possible to get debugging info from 
fscomm?


From msc at freeswitch.org  Tue Mar 16 14:14:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:14:17 -0700
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
Message-ID: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>

On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com> wrote:

> Hello,
> Pretty new, but so far very happy with freeswitch!  The setup and config is
> really nice, straightforward, and I appreciate the design of the system.
>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
> really looking for something more stable, and I think I've found it here!
>
> One issue I've run into, and I'm not sure if its just that my use case is
> abnormal, hence the email.
>
> I've worked with PBX systems for a number of years (11 now...), and in all
> of the systems I've worked with, call parking was kind of a "roaming"
> feature.  By this I mean, you think party X (the intended recipient of the
> call) is in the building, but they are not at their desk.  So, you park the
> call, and then page, or by some other means attempt to locate party X and
> tell them "get to the nearest phone and pick up 5901".  If you can't locate
> party X or you can't find them in time, the call should time out back to the
> parking party (receptionist) to be handled appropriately (message taken,
> sent to voicemail, transferred to party X's boss)
>
> This scenario requires a few things:
> 1) the receptionist/person who actually receives the call needs to be able
> to put the call in a specific place (extension) that doesn't change
> 2) Party X (the intended recipient) needs to be able to access that
> specific place (extension) from any phone inside the company
> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
> inadequate, because if the receptionist parks 2 people on 5900, one for
> party X and one for party Y, in that order, but party Y gets to a phone
> first, they will be connected to party X's caller)
> 4) The receptionist has to be protected from or provided with sufficient
> information to avoid connecting 2 customers with each other (IE the
> valet_park in freeswitch is inadequate, because if she parks a caller for X
> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
> what can she do?  What if she just forgets and sends that caller to 6001?
>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
> there...
>
> The closest I can get to implementing the above scenario is by using
> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
> receptionist phone to "notify" her that someone is already parked in a
> certain queue.  Unfortunately, there is no enforcement of this policy by the
> system, and if she fat fingers her transfer, she will double up a queue,
> creating the problem indicated in 3 above.  The other shortcoming of this
> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
> becomes impossible to appropriately handle call park.  If the receptionist
> is away from her desk and attempts to answer a call, she can't tell which
> parking spots are free at all on a regular 2 line phone.  Also, regular
> users are prohibited from using call park as they cannot know which slots
> are free (the need for this arises often in certain industries, such as law,
> client is speaking with attorney, attorney finishes and says "let me connect
> you with my assistant to schedule an appointment and handle x, y, and z for
> you.", paralegal is not at desk, atty needs to park call and page his
> assistant, maybe they are in the copy room, or the library doing
> research...)
>
> The asterisk parking lot seems to solve this problem quite elegantly
> (although the stability issues of asterisk undermine it).  It is simple to
> use (receptionist and everyone else only needs to remember 1 extension to
> park any call), The callers/parking spots are maintained by the system to
> prevent double parking, and each caller sits at a specific extension until
> they are picked up or time out.
>
> So, the question boils down to how do people using freeswitch view/use call
> park?  Is there a "better" roaming feature that I just don't know about?  If
> it is just something that hasn't been implemented/thought about this way
> yet, any idea what it would take to get a call parking feature that
> satisfies these requirements into freeswitch?
>
> Thanks for your time, sorry for the long email, hopefully it explains well
> enough what I'm trying to accomplish.
>
> -Tom
>

Tom,

First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
thank you for actually looking at the documentation before asking your
question.

You've got yourself quite the little conundrum. Many of us use park in a
more limited role, or we rely on the receptionist not to do silly things
like sending one external caller to the slot of another external caller. We
all agree that's a bad idea, unless it is done deliberately to connect two
external parties. It sounds to me like this particular situation needs an
operator panel app where the receptionist can have a pretty web page showing
what calls are parked where. To my knowledge, this does not yet exist in
FreeSWITCH. The good news is that FS provides all the requisite building
blocks to create such an application using the event socket. How do you feel
about doing a little development? :)

-MC
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From msc at freeswitch.org  Tue Mar 16 14:19:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:19:33 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
Message-ID: <87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>

On Mon, Mar 15, 2010 at 1:21 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Hello.
>
> How can I intercept a fifo? I tried to use the example of intecept
> default dialplan, but in no fifo dialed_extension.
>

Are you trying to intercept the call that is ringing at a FIFO agent's
phone?
-MC


>
>    <extension name="operators">
>      <condition field="destination_number" expression="^operators$">
>            <action application="bind_meta_app" data="2 b s
>
> record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>            <action application="hash"
>
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>            <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>            <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@
> ${domain_name}
> var callgroup)}"/>
>            <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>            <action application="answer"/>
>            <action application="playback"
> data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
>            <action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>            <action application="set" data="fifo_music=$${hold_music}"/>
>            <action application="fifo" data="operators@$${domain} in
> undef"/>
>      </condition>
>    </extension>
>
>    <fifo name="operators@$${domain}" importance="0">
>      <member timeout="15" simo="1"
> lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@
> $${domain}</member>
>    </fifo>
>  </fifos>
> --
> Best regards, Vladimir Elizarov
>
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From brian at freeswitch.org  Tue Mar 16 14:10:00 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 16:10:00 -0500
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
	<86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
Message-ID: <3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>

A is ingress to the switch.. B is egress from the switch... remember each bridged call consists of two legs.  But their are cases that can blur those lines.

/b

On Mar 16, 2010, at 3:59 PM, Troy Anderson wrote:

> Thanks, guys!  Setting the legs param to "ab" brought in lots more cdr data, including the LOSE_RACE legs.  I'm still a bit confused, though - can you explain what is considered a-leg vs. b-leg in this context?  I would have thought the call from outside into the system would constitute an a-leg.
> 
> Thanks!
> Troy



From william.suffill at gmail.com  Tue Mar 16 14:36:28 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 17:36:28 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003161411.43845.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca> 
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com> 
	<201003161411.43845.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>

With FSComm running you can use fs_cli to try pa list. If u can't list the
devices that would be the problem. I'm going to try to dig into this further
on 1 of my desktops w/ pulseaudio.

-- W
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From msc at freeswitch.org  Tue Mar 16 16:06:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 16:06:49 -0700
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
	<86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
	<3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>
Message-ID: <87f2f3b91003161606s1c1a3996t65a9ddbbd464e010@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:10 PM, Brian West <brian at freeswitch.org> wrote:

> A is ingress to the switch.. B is egress from the switch... remember each
> bridged call consists of two legs.  But their are cases that can blur those
> lines.
>
> FYI, this is a good piece of information. I added it explicitly to the call
legs wiki page that we just started:
http://wiki.freeswitch.org/wiki/Call_Legs

If anyone else has something to add by all means do so.
-MC
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From jmesquita at freeswitch.org  Tue Mar 16 16:13:26 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 16 Mar 2010 20:13:26 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca>
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
Message-ID: <ebbc2d551003161613g15bfc624h112d446ee68f1716@mail.gmail.com>

Thanks for the help W!

Jo?o Mesquita


On Tue, Mar 16, 2010 at 6:36 PM, William Suffill
<william.suffill at gmail.com>wrote:

> With FSComm running you can use fs_cli to try pa list. If u can't list the
> devices that would be the problem. I'm going to try to dig into this further
> on 1 of my desktops w/ pulseaudio.
>
> -- W
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 16:55:56 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 19:55:56 -0400
Subject: [Freeswitch-users] MWI Question
Message-ID: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>

Hi All,

I delete all the voice mails in the voice mail folder for a particular
extension.

But the phone[AASTRA 9133i]  MWI keep on blinking and and display showing
the no of messages.

How can I fix this issue.

Thanks
Lloyd
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From delorenzodesign at gmail.com  Tue Mar 16 17:23:27 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 20:23:27 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
Message-ID: <5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>

So I tried Michael's suggestion of:

digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
"", "\\d+");

but it still didn't detect that I had pressed any keys.  What would cause
this?  Is there someplace I can check to see if anything at all is being
detected?


On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> I'm having some trouble getting consistent results with playAndGetDigits,
>> sometimes the digits are record and other times they're not.  It seemed to
>> have something to do with how quickly the digits were pressed, but now that
>> no longer seems to be the case.
>>
>> I'm trying to have users press a confirmation code, in this case "1111"
>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>> possible.
>>
>> Can anyone point me in the right direction?
>>
>>
>>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
>> .. provider .. "/1" .. number_to_call);
>>
>>     while(session:ready()) do
>>        session:answer();
>>        session:setInputCallback("onInput","true");
>>
>>        --playbackMessage(check_message);
>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>> get_digits_message, "", "[1]");
>>
>
> You are specifying a minimum of four and a maximum of four digits. Also,
> your regex will match any string that has a digit 1 in it, which may or may
> not be what you are looking for. Lastly, you haven't specified a terminator
> key (like "#") but you are giving the user only one try and waiting 5000ms
> for the entry. Try something like this:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
> "", "\\d+");
>
> Let us know if that works. Oh, one other thing: you might want to add an
> invalid message because playAndGetDigits will handle invalid input for you.
> -MC
>
> P.S. - This topic is covered in greater detail in Chapter 7 of the upcoming
> FreeSWITCH book. ;)
>
>
>
>>        freeswitch.consoleLog("info", digits .. "\n");
>>        freeswitch.consoleLog("info", "Did this check out? " ..
>> (check_message == true and "yes" or "no" .. "\n"));
>>
>>        -- we got what we're looking for, so we can end this
>>        if(check_success) then
>>           freeswitch.consoleLog("info", "A positive response was received
>> from this check.\n");
>>           break;
>>        else
>>           -- not a positive response
>>           -- if we reached our max attempts, we're finished move on
>>           if(attempts == max_attempts) then
>>             freeswitch.consoleLog("info", "We've reached our maximum
>> attempts for this number.\n");
>>             break;
>>           end
>>        end
>>     end
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 17:37:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 20:37:38 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
Message-ID: <8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>

set the console log level to 7. Then you can see the DTMF values in console
log , while you are pressing the keys.

Lloyd

On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> So I tried Michael's suggestion of:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
> "", "\\d+");
>
> but it still didn't detect that I had pressed any keys.  What would cause
> this?  Is there someplace I can check to see if anything at all is being
> detected?
>
>
> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> I'm having some trouble getting consistent results with playAndGetDigits,
>>> sometimes the digits are record and other times they're not.  It seemed to
>>> have something to do with how quickly the digits were pressed, but now that
>>> no longer seems to be the case.
>>>
>>> I'm trying to have users press a confirmation code, in this case "1111"
>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>> possible.
>>>
>>> Can anyone point me in the right direction?
>>>
>>>
>>>     session =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>> "/1" .. number_to_call);
>>>
>>>     while(session:ready()) do
>>>        session:answer();
>>>        session:setInputCallback("onInput","true");
>>>
>>>        --playbackMessage(check_message);
>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>> get_digits_message, "", "[1]");
>>>
>>
>> You are specifying a minimum of four and a maximum of four digits. Also,
>> your regex will match any string that has a digit 1 in it, which may or may
>> not be what you are looking for. Lastly, you haven't specified a terminator
>> key (like "#") but you are giving the user only one try and waiting 5000ms
>> for the entry. Try something like this:
>>
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>> "", "\\d+");
>>
>> Let us know if that works. Oh, one other thing: you might want to add an
>> invalid message because playAndGetDigits will handle invalid input for you.
>> -MC
>>
>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>> upcoming FreeSWITCH book. ;)
>>
>>
>>
>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>> (check_message == true and "yes" or "no" .. "\n"));
>>>
>>>        -- we got what we're looking for, so we can end this
>>>        if(check_success) then
>>>           freeswitch.consoleLog("info", "A positive response was received
>>> from this check.\n");
>>>           break;
>>>        else
>>>           -- not a positive response
>>>           -- if we reached our max attempts, we're finished move on
>>>           if(attempts == max_attempts) then
>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>> attempts for this number.\n");
>>>             break;
>>>           end
>>>        end
>>>     end
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 17:43:31 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 20:43:31 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
Message-ID: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>

when a user parked to a slot how to handle the park time out. Is there any
way to transfer to the call back to a receptionist.

Thanks
Lloyd



On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>
>> Hello,
>> Pretty new, but so far very happy with freeswitch!  The setup and config
>> is really nice, straightforward, and I appreciate the design of the system.
>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>> really looking for something more stable, and I think I've found it here!
>>
>> One issue I've run into, and I'm not sure if its just that my use case is
>> abnormal, hence the email.
>>
>> I've worked with PBX systems for a number of years (11 now...), and in all
>> of the systems I've worked with, call parking was kind of a "roaming"
>> feature.  By this I mean, you think party X (the intended recipient of the
>> call) is in the building, but they are not at their desk.  So, you park the
>> call, and then page, or by some other means attempt to locate party X and
>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>> party X or you can't find them in time, the call should time out back to the
>> parking party (receptionist) to be handled appropriately (message taken,
>> sent to voicemail, transferred to party X's boss)
>>
>> This scenario requires a few things:
>> 1) the receptionist/person who actually receives the call needs to be able
>> to put the call in a specific place (extension) that doesn't change
>> 2) Party X (the intended recipient) needs to be able to access that
>> specific place (extension) from any phone inside the company
>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>> inadequate, because if the receptionist parks 2 people on 5900, one for
>> party X and one for party Y, in that order, but party Y gets to a phone
>> first, they will be connected to party X's caller)
>> 4) The receptionist has to be protected from or provided with sufficient
>> information to avoid connecting 2 customers with each other (IE the
>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>> what can she do?  What if she just forgets and sends that caller to 6001?
>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>> there...
>>
>> The closest I can get to implementing the above scenario is by using
>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>> receptionist phone to "notify" her that someone is already parked in a
>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>> system, and if she fat fingers her transfer, she will double up a queue,
>> creating the problem indicated in 3 above.  The other shortcoming of this
>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>> becomes impossible to appropriately handle call park.  If the receptionist
>> is away from her desk and attempts to answer a call, she can't tell which
>> parking spots are free at all on a regular 2 line phone.  Also, regular
>> users are prohibited from using call park as they cannot know which slots
>> are free (the need for this arises often in certain industries, such as law,
>> client is speaking with attorney, attorney finishes and says "let me connect
>> you with my assistant to schedule an appointment and handle x, y, and z for
>> you.", paralegal is not at desk, atty needs to park call and page his
>> assistant, maybe they are in the copy room, or the library doing
>> research...)
>>
>> The asterisk parking lot seems to solve this problem quite elegantly
>> (although the stability issues of asterisk undermine it).  It is simple to
>> use (receptionist and everyone else only needs to remember 1 extension to
>> park any call), The callers/parking spots are maintained by the system to
>> prevent double parking, and each caller sits at a specific extension until
>> they are picked up or time out.
>>
>> So, the question boils down to how do people using freeswitch view/use
>> call park?  Is there a "better" roaming feature that I just don't know
>> about?  If it is just something that hasn't been implemented/thought about
>> this way yet, any idea what it would take to get a call parking feature that
>> satisfies these requirements into freeswitch?
>>
>> Thanks for your time, sorry for the long email, hopefully it explains well
>> enough what I'm trying to accomplish.
>>
>> -Tom
>>
>
> Tom,
>
> First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
> thank you for actually looking at the documentation before asking your
> question.
>
> You've got yourself quite the little conundrum. Many of us use park in a
> more limited role, or we rely on the receptionist not to do silly things
> like sending one external caller to the slot of another external caller. We
> all agree that's a bad idea, unless it is done deliberately to connect two
> external parties. It sounds to me like this particular situation needs an
> operator panel app where the receptionist can have a pretty web page showing
> what calls are parked where. To my knowledge, this does not yet exist in
> FreeSWITCH. The good news is that FS provides all the requisite building
> blocks to create such an application using the event socket. How do you feel
> about doing a little development? :)
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From delorenzodesign at gmail.com  Tue Mar 16 17:51:38 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 20:51:38 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
Message-ID: <5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>

Ok, I updated the console log level, and there are no DTMF values being
detected:

4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
160 bytes per 20ms
2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
sofia/external/19736327407!
2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
sofia/external/19736327407 entering state [completing][200]
2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
v=0
o=root 21648 21649 IN IP4 74.51.38.156
s=session
c=IN IP4 74.51.38.156
t=0 0
m=audio 15150 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
sofia/external/19736327407 entering state [ready][200]
2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
[sofia/external/19736327407] has been answered
2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
Resulted in Success: [sofia/external/19736327407]
2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
(sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
CS_SOFT_EXECUTE
2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
sofia/external/19736327407 Standard SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE going to sleep
2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
confirmed.
2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
file
2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received during
call from playAndGetDigits:
2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out? no

On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> set the console log level to 7. Then you can see the DTMF values in console
> log , while you are pressing the keys.
>
> Lloyd
>
>
> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> So I tried Michael's suggestion of:
>>
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>> "", "\\d+");
>>
>> but it still didn't detect that I had pressed any keys.  What would cause
>> this?  Is there someplace I can check to see if anything at all is being
>> detected?
>>
>>
>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> I'm having some trouble getting consistent results with
>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>> not.  It seemed to have something to do with how quickly the digits were
>>>> pressed, but now that no longer seems to be the case.
>>>>
>>>> I'm trying to have users press a confirmation code, in this case "1111"
>>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>> possible.
>>>>
>>>> Can anyone point me in the right direction?
>>>>
>>>>
>>>>     session =
>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>> "/1" .. number_to_call);
>>>>
>>>>     while(session:ready()) do
>>>>        session:answer();
>>>>        session:setInputCallback("onInput","true");
>>>>
>>>>        --playbackMessage(check_message);
>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>> get_digits_message, "", "[1]");
>>>>
>>>
>>> You are specifying a minimum of four and a maximum of four digits. Also,
>>> your regex will match any string that has a digit 1 in it, which may or may
>>> not be what you are looking for. Lastly, you haven't specified a terminator
>>> key (like "#") but you are giving the user only one try and waiting 5000ms
>>> for the entry. Try something like this:
>>>
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>>> "", "\\d+");
>>>
>>> Let us know if that works. Oh, one other thing: you might want to add an
>>> invalid message because playAndGetDigits will handle invalid input for you.
>>> -MC
>>>
>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>> upcoming FreeSWITCH book. ;)
>>>
>>>
>>>
>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>
>>>>        -- we got what we're looking for, so we can end this
>>>>        if(check_success) then
>>>>           freeswitch.consoleLog("info", "A positive response was
>>>> received from this check.\n");
>>>>           break;
>>>>        else
>>>>           -- not a positive response
>>>>           -- if we reached our max attempts, we're finished move on
>>>>           if(attempts == max_attempts) then
>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>> attempts for this number.\n");
>>>>             break;
>>>>           end
>>>>        end
>>>>     end
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 18:07:15 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 21:07:15 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
Message-ID: <8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>

here is the test script I used

session:answer();
digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
"", "\\d+");

Here is the console log

2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
*2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800*
*2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640*
*2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800*
*2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640*
2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
[1234][\d+]

You may have some issue with the provider. Check using a local extension.


On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Ok, I updated the console log level, and there are no DTMF values being
> detected:
>
> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
> 160 bytes per 20ms
> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
> sofia/external/19736327407!
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
> sofia/external/19736327407 entering state [completing][200]
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
> v=0
> o=root 21648 21649 IN IP4 74.51.38.156
> s=session
> c=IN IP4 74.51.38.156
> t=0 0
> m=audio 15150 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
> sofia/external/19736327407 entering state [ready][200]
> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
> [sofia/external/19736327407] has been answered
> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
> Resulted in Success: [sofia/external/19736327407]
> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
> CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/19736327407 [BREAK]
> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
> (sofia/external/19736327407) State SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
> sofia/external/19736327407 Standard SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
> confirmed.
> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
> file
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
> during call from playAndGetDigits:
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
> no
>
>
> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> set the console log level to 7. Then you can see the DTMF values in
>> console log , while you are pressing the keys.
>>
>> Lloyd
>>
>>
>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> So I tried Michael's suggestion of:
>>>
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>>> "", "\\d+");
>>>
>>> but it still didn't detect that I had pressed any keys.  What would cause
>>> this?  Is there someplace I can check to see if anything at all is being
>>> detected?
>>>
>>>
>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> I'm having some trouble getting consistent results with
>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>> pressed, but now that no longer seems to be the case.
>>>>>
>>>>> I'm trying to have users press a confirmation code, in this case "1111"
>>>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>> possible.
>>>>>
>>>>> Can anyone point me in the right direction?
>>>>>
>>>>>
>>>>>     session =
>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>> "/1" .. number_to_call);
>>>>>
>>>>>     while(session:ready()) do
>>>>>        session:answer();
>>>>>        session:setInputCallback("onInput","true");
>>>>>
>>>>>        --playbackMessage(check_message);
>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>> get_digits_message, "", "[1]");
>>>>>
>>>>
>>>> You are specifying a minimum of four and a maximum of four digits. Also,
>>>> your regex will match any string that has a digit 1 in it, which may or may
>>>> not be what you are looking for. Lastly, you haven't specified a terminator
>>>> key (like "#") but you are giving the user only one try and waiting 5000ms
>>>> for the entry. Try something like this:
>>>>
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> get_digits_message, "", "\\d+");
>>>>
>>>> Let us know if that works. Oh, one other thing: you might want to add an
>>>> invalid message because playAndGetDigits will handle invalid input for you.
>>>> -MC
>>>>
>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>> upcoming FreeSWITCH book. ;)
>>>>
>>>>
>>>>
>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>
>>>>>        -- we got what we're looking for, so we can end this
>>>>>        if(check_success) then
>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>> received from this check.\n");
>>>>>           break;
>>>>>        else
>>>>>           -- not a positive response
>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>           if(attempts == max_attempts) then
>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>> attempts for this number.\n");
>>>>>             break;
>>>>>           end
>>>>>        end
>>>>>     end
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Tue Mar 16 18:08:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 20:08:32 -0500
Subject: [Freeswitch-users] MWI Question
In-Reply-To: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>
References: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>
Message-ID: <b39273531003161808y6cf3fe8aqda1d68421cf33c2f@mail.gmail.com>

Next time use vm_delete rather than removing the files.  This may still work
for you.

"vm_delete id at domain" should work in your case.

If it doesn't, connect to the voicemail database and issue:

delete from voicemail_msgs where username='id' and domain='domain';

On Tue, Mar 16, 2010 at 6:55 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I delete all the voice mails in the voice mail folder for a particular
> extension.
>
> But the phone[AASTRA 9133i]  MWI keep on blinking and and display showing
> the no of messages.
>
> How can I fix this issue.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Tue Mar 16 18:10:29 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 20:10:29 -0500
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
Message-ID: <b39273531003161810m521cc9a0mba8aa9de103fcf03@mail.gmail.com>

I would think you could use:

sched_transfer,Schedule a transfer in the future,[+]<time> <extension>
<dialplan> <context>,mod_dptools

before transfering to the slot.  I haven't tried it, but don't see why it
wouldn't work.  More docs on the wiki.

On Tue, Mar 16, 2010 at 7:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> when a user parked to a slot how to handle the park time out. Is there any
> way to transfer to the call back to a receptionist.
>
> Thanks
> Lloyd
>
>
>
> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From delorenzodesign at gmail.com  Tue Mar 16 18:16:10 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 21:16:10 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
Message-ID: <5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>

What do you mean about the provider?  I'm not sure I understand that part.
I'm trying to call a number and have the call's recipient type the digits
in.

On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> here is the test script I used
>
> session:answer();
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
> "", "\\d+");
>
> Here is the console log
>
> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800*
> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640*
> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800*
> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640*
> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
> [1234][\d+]
>
> You may have some issue with the provider. Check using a local extension.
>
>
> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> Ok, I updated the console log level, and there are no DTMF values being
>> detected:
>>
>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
>> 160 bytes per 20ms
>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>> sofia/external/19736327407!
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>> sofia/external/19736327407 entering state [completing][200]
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>> v=0
>> o=root 21648 21649 IN IP4 74.51.38.156
>> s=session
>> c=IN IP4 74.51.38.156
>> t=0 0
>> m=audio 15150 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>>
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>> sofia/external/19736327407 entering state [ready][200]
>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>> [sofia/external/19736327407] has been answered
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>> Resulted in Success: [sofia/external/19736327407]
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>> CS_SOFT_EXECUTE
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/external/19736327407 [BREAK]
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>> (sofia/external/19736327407) State SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>> sofia/external/19736327407 Standard SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>> confirmed.
>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
>> file
>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>> during call from playAndGetDigits:
>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
>> no
>>
>>
>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> set the console log level to 7. Then you can see the DTMF values in
>>> console log , while you are pressing the keys.
>>>
>>> Lloyd
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> So I tried Michael's suggestion of:
>>>>
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> get_digits_message, "", "\\d+");
>>>>
>>>> but it still didn't detect that I had pressed any keys.  What would
>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>> being detected?
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> I'm having some trouble getting consistent results with
>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>
>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>> possible.
>>>>>>
>>>>>> Can anyone point me in the right direction?
>>>>>>
>>>>>>
>>>>>>     session =
>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>> "/1" .. number_to_call);
>>>>>>
>>>>>>     while(session:ready()) do
>>>>>>        session:answer();
>>>>>>        session:setInputCallback("onInput","true");
>>>>>>
>>>>>>        --playbackMessage(check_message);
>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>> get_digits_message, "", "[1]");
>>>>>>
>>>>>
>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> get_digits_message, "", "\\d+");
>>>>>
>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>> you.
>>>>> -MC
>>>>>
>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>> upcoming FreeSWITCH book. ;)
>>>>>
>>>>>
>>>>>
>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>
>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>        if(check_success) then
>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>> received from this check.\n");
>>>>>>           break;
>>>>>>        else
>>>>>>           -- not a positive response
>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>           if(attempts == max_attempts) then
>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>> attempts for this number.\n");
>>>>>>             break;
>>>>>>           end
>>>>>>        end
>>>>>>     end
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 18:34:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 21:34:46 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
Message-ID: <8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>

You inbound DID Provider


On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> What do you mean about the provider?  I'm not sure I understand that part.
> I'm trying to call a number and have the call's recipient type the digits
> in.
>
>
> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> here is the test script I used
>>
>> session:answer();
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
>> "", "\\d+");
>>
>> Here is the console log
>>
>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
>> file
>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800
>> *
>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640
>> *
>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800
>> *
>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640
>> *
>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>> [1234][\d+]
>>
>> You may have some issue with the provider. Check using a local extension.
>>
>>
>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> Ok, I updated the console log level, and there are no DTMF values being
>>> detected:
>>>
>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>> sofia/external/19736327407!
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>> sofia/external/19736327407 entering state [completing][200]
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>> v=0
>>> o=root 21648 21649 IN IP4 74.51.38.156
>>> s=session
>>> c=IN IP4 74.51.38.156
>>> t=0 0
>>> m=audio 15150 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>> sofia/external/19736327407 entering state [ready][200]
>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>> [sofia/external/19736327407] has been answered
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>>> Resulted in Success: [sofia/external/19736327407]
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>> CS_SOFT_EXECUTE
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/external/19736327407 [BREAK]
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>> confirmed.
>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>> playing file
>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>> during call from playAndGetDigits:
>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
>>> no
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> set the console log level to 7. Then you can see the DTMF values in
>>>> console log , while you are pressing the keys.
>>>>
>>>> Lloyd
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> So I tried Michael's suggestion of:
>>>>>
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> get_digits_message, "", "\\d+");
>>>>>
>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>> being detected?
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>
>>>>>>> I'm having some trouble getting consistent results with
>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>
>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>> possible.
>>>>>>>
>>>>>>> Can anyone point me in the right direction?
>>>>>>>
>>>>>>>
>>>>>>>     session =
>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>> "/1" .. number_to_call);
>>>>>>>
>>>>>>>     while(session:ready()) do
>>>>>>>        session:answer();
>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>
>>>>>>>        --playbackMessage(check_message);
>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>> get_digits_message, "", "[1]");
>>>>>>>
>>>>>>
>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>
>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>> get_digits_message, "", "\\d+");
>>>>>>
>>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>>> you.
>>>>>> -MC
>>>>>>
>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>
>>>>>>
>>>>>>
>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>
>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>        if(check_success) then
>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>> received from this check.\n");
>>>>>>>           break;
>>>>>>>        else
>>>>>>>           -- not a positive response
>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>           if(attempts == max_attempts) then
>>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>>> attempts for this number.\n");
>>>>>>>             break;
>>>>>>>           end
>>>>>>>        end
>>>>>>>     end
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From delorenzodesign at gmail.com  Tue Mar 16 19:05:01 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 22:05:01 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
Message-ID: <5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>

So I'm making calls using VoiceNetwork, are you suggesting there's a problem
there?  Or on the line where the call is received (I've tried on a couple of
different networks/providers)?

On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> You inbound DID Provider
>
>
>
> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> What do you mean about the provider?  I'm not sure I understand that
>> part.  I'm trying to call a number and have the call's recipient type the
>> digits in.
>>
>>
>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> here is the test script I used
>>>
>>> session:answer();
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
>>> "", "\\d+");
>>>
>>> Here is the console log
>>>
>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>> playing file
>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 1:800*
>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 2:640*
>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 3:800*
>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 4:640*
>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>>> [1234][\d+]
>>>
>>> You may have some issue with the provider. Check using a local extension.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> Ok, I updated the console log level, and there are no DTMF values being
>>>> detected:
>>>>
>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>> sofia/external/19736327407!
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>> sofia/external/19736327407 entering state [completing][200]
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>> v=0
>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>> s=session
>>>> c=IN IP4 74.51.38.156
>>>> t=0 0
>>>> m=audio 15150 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>> sofia/external/19736327407 entering state [ready][200]
>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>> [sofia/external/19736327407] has been answered
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>>>> Resulted in Success: [sofia/external/19736327407]
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>> CS_SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/external/19736327407 [BREAK]
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>> Activated L16 at 8000hz 1 channels 20ms
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>> confirmed.
>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>> playing file
>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>> during call from playAndGetDigits:
>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>> out? no
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <
>>>> lloyd.aloysius at gmail.com> wrote:
>>>>
>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>> console log , while you are pressing the keys.
>>>>>
>>>>> Lloyd
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> So I tried Michael's suggestion of:
>>>>>>
>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>> get_digits_message, "", "\\d+");
>>>>>>
>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>> being detected?
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>
>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>
>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>> possible.
>>>>>>>>
>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>
>>>>>>>>
>>>>>>>>     session =
>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>> "/1" .. number_to_call);
>>>>>>>>
>>>>>>>>     while(session:ready()) do
>>>>>>>>        session:answer();
>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>
>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>
>>>>>>>
>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>
>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>
>>>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>>>> you.
>>>>>>> -MC
>>>>>>>
>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>
>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>        if(check_success) then
>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>> received from this check.\n");
>>>>>>>>           break;
>>>>>>>>        else
>>>>>>>>           -- not a positive response
>>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>>>> attempts for this number.\n");
>>>>>>>>             break;
>>>>>>>>           end
>>>>>>>>        end
>>>>>>>>     end
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pavera at gmail.com  Tue Mar 16 19:17:13 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 20:17:13 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
Message-ID: <82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>

On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>
>> Hello,
>> Pretty new, but so far very happy with freeswitch!  The setup and config
>> is really nice, straightforward, and I appreciate the design of the system.
>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>> really looking for something more stable, and I think I've found it here!
>>
>> One issue I've run into, and I'm not sure if its just that my use case is
>> abnormal, hence the email.
>>
>> I've worked with PBX systems for a number of years (11 now...), and in all
>> of the systems I've worked with, call parking was kind of a "roaming"
>> feature.  By this I mean, you think party X (the intended recipient of the
>> call) is in the building, but they are not at their desk.  So, you park the
>> call, and then page, or by some other means attempt to locate party X and
>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>> party X or you can't find them in time, the call should time out back to the
>> parking party (receptionist) to be handled appropriately (message taken,
>> sent to voicemail, transferred to party X's boss)
>>
>> This scenario requires a few things:
>> 1) the receptionist/person who actually receives the call needs to be able
>> to put the call in a specific place (extension) that doesn't change
>> 2) Party X (the intended recipient) needs to be able to access that
>> specific place (extension) from any phone inside the company
>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>> inadequate, because if the receptionist parks 2 people on 5900, one for
>> party X and one for party Y, in that order, but party Y gets to a phone
>> first, they will be connected to party X's caller)
>> 4) The receptionist has to be protected from or provided with sufficient
>> information to avoid connecting 2 customers with each other (IE the
>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>> what can she do?  What if she just forgets and sends that caller to 6001?
>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>> there...
>>
>> The closest I can get to implementing the above scenario is by using
>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>> receptionist phone to "notify" her that someone is already parked in a
>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>> system, and if she fat fingers her transfer, she will double up a queue,
>> creating the problem indicated in 3 above.  The other shortcoming of this
>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>> becomes impossible to appropriately handle call park.  If the receptionist
>> is away from her desk and attempts to answer a call, she can't tell which
>> parking spots are free at all on a regular 2 line phone.  Also, regular
>> users are prohibited from using call park as they cannot know which slots
>> are free (the need for this arises often in certain industries, such as law,
>> client is speaking with attorney, attorney finishes and says "let me connect
>> you with my assistant to schedule an appointment and handle x, y, and z for
>> you.", paralegal is not at desk, atty needs to park call and page his
>> assistant, maybe they are in the copy room, or the library doing
>> research...)
>>
>> The asterisk parking lot seems to solve this problem quite elegantly
>> (although the stability issues of asterisk undermine it).  It is simple to
>> use (receptionist and everyone else only needs to remember 1 extension to
>> park any call), The callers/parking spots are maintained by the system to
>> prevent double parking, and each caller sits at a specific extension until
>> they are picked up or time out.
>>
>> So, the question boils down to how do people using freeswitch view/use
>> call park?  Is there a "better" roaming feature that I just don't know
>> about?  If it is just something that hasn't been implemented/thought about
>> this way yet, any idea what it would take to get a call parking feature that
>> satisfies these requirements into freeswitch?
>>
>> Thanks for your time, sorry for the long email, hopefully it explains well
>> enough what I'm trying to accomplish.
>>
>> -Tom
>>
>
> Tom,
>
> First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
> thank you for actually looking at the documentation before asking your
> question.
>
> You've got yourself quite the little conundrum. Many of us use park in a
> more limited role, or we rely on the receptionist not to do silly things
> like sending one external caller to the slot of another external caller. We
> all agree that's a bad idea, unless it is done deliberately to connect two
> external parties. It sounds to me like this particular situation needs an
> operator panel app where the receptionist can have a pretty web page showing
> what calls are parked where. To my knowledge, this does not yet exist in
> FreeSWITCH. The good news is that FS provides all the requisite building
> blocks to create such an application using the event socket. How do you feel
> about doing a little development? :)
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Michael,
Thanks for the reply. I worked on it more with the business in question
today and we came up with a hybrid solution which I think will cover all the
bases.
1) I implemented a "call park" extension on each phone, these are standard
mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
freeswitch config here BTW, like 8 lines in the dialplan handles this for
all 50 extensions).  This way everyone has a call park extension that can be
picked up anywhere, the numbering scheme is essentially take your extension
prepend a 2, that is your call park extension, so if someone pages you, you
just pick up any phone and dial 2+ext and that will pick the call up.
 Everyone in the office can use this feature to park a call for anyone,
because only calls for a specific person will ever be parked in that queue,
you can put multiple people there without issue.
2) Implemented as mentioned a "parking lot" set of mod_fifo queues that have
SLA/BLA set up on the receptionist phone.  She can use these queues to
quickly and easily park a bunch of people at once (IE when there are call
floods) and then send them off to where they need to go as time permits.  If
she isn't at her phone, she can still answer calls and park them onto
individual users park extensions.  Still doesn't have the "safety" of not
double parking someone in the parking lot, but we'll trust her to be able to
see red lights :)

This whole thing goes live Thursday, so wish me luck :)
Thanks again!
Tom
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From pavera at gmail.com  Tue Mar 16 19:18:46 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 20:18:46 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
Message-ID: <82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_fifo#Dialplan_Example
At least with mod_fifo (the call parking solution I'm using) there is a
pretty clear example in the wiki.

On Tue, Mar 16, 2010 at 6:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> when a user parked to a slot how to handle the park time out. Is there any
> way to transfer to the call back to a receptionist.
>
> Thanks
> Lloyd
>
>
>
> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 16 20:38:58 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Mar 2010 20:38:58 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
Message-ID: <D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>

Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
-MC

Sent from my iPhone

On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
 > wrote:

> So I'm making calls using VoiceNetwork, are you suggesting there's a  
> problem there?  Or on the line where the call is received (I've  
> tried on a couple of different networks/providers)?
>
> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> You inbound DID Provider
>
>
>
> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> What do you mean about the provider?  I'm not sure I understand that  
> part.  I'm trying to call a number and have the call's recipient  
> type the digits in.
>
>
> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> here is the test script I used
>
> session:answer();
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter- 
> digits.wav", "", "\\d+");
>
> Here is the console log
>
> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 1:800
> 2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:640
> 2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 3:800
> 2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 4:640
> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test  
> Regex [1234][\d+]
>
> You may have some issue with the provider. Check using a local  
> extension.
>
>
> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> Ok, I updated the console log level, and there are no DTMF values  
> being detected:
>
> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer  
> [soft] 160 bytes per 20ms
> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer  
> sofia/external/19736327407!
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel sofia/ 
> external/19736327407 entering state [completing][200]
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
> v=0
> o=root 21648 21649 IN IP4 74.51.38.156
> s=session
> c=IN IP4 74.51.38.156
> t=0 0
> m=audio 15150 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel sofia/ 
> external/19736327407 entering state [ready][200]
> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel [sofia/ 
> external/19736327407] has been answered
> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107  
> Originate Resulted in Success: [sofia/external/19736327407]
> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497 (sofia/ 
> external/19736327407) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send  
> signal sofia/external/19736327407 [BREAK]
> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354  
> (sofia/external/19736327407) State SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200  
> sofia/external/19736327407 Standard SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354  
> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port  
> confirmed.
> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done  
> playing file
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits  
> received during call from playAndGetDigits:
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check  
> out? no
>
>
> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> set the console log level to 7. Then you can see the DTMF values in  
> console log , while you are pressing the keys.
>
> Lloyd
>
>
> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> So I tried Michael's suggestion of:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",  
> get_digits_message, "", "\\d+");
>
> but it still didn't detect that I had pressed any keys.  What would  
> cause this?  Is there someplace I can check to see if anything at  
> all is being detected?
>
>
> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> I'm having some trouble getting consistent results with  
> playAndGetDigits, sometimes the digits are record and other times  
> they're not.  It seemed to have something to do with how quickly the  
> digits were pressed, but now that no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case  
> "1111" although I'd like to make it accept "1" or "11" or "111" or  
> "1111" if possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/ 
> gateway/" .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",  
> get_digits_message, "", "[1]");
>
> You are specifying a minimum of four and a maximum of four digits.  
> Also, your regex will match any string that has a digit 1 in it,  
> which may or may not be what you are looking for. Lastly, you  
> haven't specified a terminator key (like "#") but you are giving the  
> user only one try and waiting 5000ms for the entry. Try something  
> like this:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",  
> get_digits_message, "", "\\d+");
>
> Let us know if that works. Oh, one other thing: you might want to  
> add an invalid message because playAndGetDigits will handle invalid  
> input for you.
> -MC
>
> P.S. - This topic is covered in greater detail in Chapter 7 of the  
> upcoming FreeSWITCH book. ;)
>
>
>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..  
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was  
> received from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum  
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
>
>
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From infos at madovsky.org  Tue Mar 16 21:20:15 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 00:20:15 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
Message-ID: <3E1709744536417C877AB5952395DAED@MOBILEE1705>

Also Reddy,

FS is protected (correct me if I'm wrong)
under MPL licence
http://www.mozilla.org/MPL/MPL-1.1.html

in the article 3, paragraph 2:
3.2. Availability of Source Code.
Any Modification which You create or to which You contribute must be made available in Source Code form under the terms of this License either on the same media as an Executable version or via an accepted Electronic Distribution Mechanism to anyone to whom you made an Executable version available; and if made available via Electronic Distribution Mechanism, must remain available for at least twelve (12) months after the date it initially became available, or at least six (6) months after a subsequent version of that particular Modification has been made available to such recipients. You are responsible for ensuring that the Source Code version remains available even if the Electronic Distribution Mechanism is maintained by a third party. 

so my opinion is you are infringing the Licence 

Regards

Franck

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:00 PM
  Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


  so share your customized version to the community and maybe
  we ll be glad to help you.... this is the principle of open source
    ----- Original Message ----- 
    From: srinivasula reddy 
    To: freeswitch-users 
    Sent: Tuesday, March 16, 2010 11:49 AM
    Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


    we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.

    srinivas


    On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:

      Hi,

      i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. 
      <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>
      i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea

      -- 
      Srinivasula Reddy K




    -- 
    Srinivasula Reddy K



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


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From jaybinks at gmail.com  Tue Mar 16 21:30:26 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 17 Mar 2010 14:30:26 +1000
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
Message-ID: <ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>

Im not on the FS dev team, but I think you may be a little mistaken.

generally your free to make modifications to open source code ( for
internal use )
however its when you distribute the resultant product, then you have
to make the source available..

if your only using it in-house, I dont believe your required to
contribute the code back.
( however your going to end up having issues maintaining your patch
set over a long period.. like can be seen here )

see here :

> must be made available in Source Code form under the terms of this License either on the
> same media as an Executable version

if your not distributing your executable to any 3rd parties... then I
dont believe you have to distribute the modified source..

Correct me if Im wrong though guys...

Jay


From maciej.aniserowicz at gmail.com  Wed Mar 17 00:39:40 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 16 Mar 2010 23:39:40 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
Message-ID: <1268811580808-4748784.post@n2.nabble.com>


Hi again,
I don't quite understand why both of them would not be parked... in logs
(from http://pastebin.freeswitch.org/12432) I can see this:

dialing 1st call
[5] 2010-03-15 08:52:45.191854 [NOTICE] switch_channel.c:666 New Channel
sofia/external/2500 [b278ec65-6d83-4454-b885-7ee66f414230]

then...:
[55] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[56] EXECUTE sofia/external/2500 park()

then...:
[57] 2010-03-15 08:52:45.332056 [DEBUG] switch_ivr.c:1426
(sofia/external/2500) State Change CS_EXECUTE -> CS_ROUTING
[58] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_session.c:1018 Send
signal sofia/external/2500 [BREAK]
[59] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_session.c:638 Send
signal sofia/external/2500 [BREAK]
[60] 2010-03-15 08:52:45.332056 [NOTICE] switch_ivr.c:1432 Transfer
sofia/external/2500 to inline[park at default]

then PARK again...:
[72] 2010-03-15 08:52:45.342070 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[73] EXECUTE sofia/external/2500 park()
[74] 2010-03-15 08:52:45.402157 [DEBUG] switch_rtp.c:2055 Correct ip/port
confirmed.
[75] 2010-03-15 08:52:50.709789 [DEBUG] switch_ivr.c:1426
(sofia/external/2500) State Change CS_EXECUTE -> CS_ROUTING

then TRANSFER again...:
[78] 2010-03-15 08:52:50.709789 [NOTICE] switch_ivr.c:1432 Transfer
sofia/external/2500 to inline[park at default]

and PARK again...:
[90] 2010-03-15 08:52:50.719803 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[91] EXECUTE sofia/external/2500 park()


And the same goes for the 2nd channel. The Transfer/Park instructions are
repeated several times. Is this normal? My command is fine according to WIKI
I guess:

bgapi originate
{originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
&park()

Or maybe the &park() is not necessary there?

This mechanism worked mostly OK for several months with previous builds of
freeswitch, the problems showed up after upgrading to trunk recently.
Sometimes it works though, i'd say that ~30% of the calls end up being
hanged up, but most of them are just fine.

Anyway, I'll try to wait for CHANNEL_PARK event as you suggested and only
then assume that the call is really "answered" and ready for processing.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4748784.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From robert.dyck at shaw.ca  Wed Mar 17 00:49:05 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 00:49:05 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
Message-ID: <201003170049.06565.robert.dyck@shaw.ca>

Is there something wrong with my build?

fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket 
Connection Error]" with fscomm running.

RD

On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> With FSComm running you can use fs_cli to try pa list. If u can't list the
> devices that would be the problem. I'm going to try to dig into this
> further on 1 of my desktops w/ pulseaudio.
> 
> -- W


From brian at microcomaustralia.com.au  Wed Mar 17 01:17:18 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 17 Mar 2010 19:17:18 +1100
Subject: [Freeswitch-users] dahdi problems
Message-ID: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>

Hello,

Just noticed zaptel was renamed dahdi.

What am I doing wrong?

2010-03-17 19:11:32.505233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/modules.conf.
2010-03-17 19:11:32.506233 [NOTICE] zap_io.c:3048 Modules configured: 1
2010-03-17 19:11:32.506233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/openzap.conf.
2010-03-17 19:11:32.506233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.506233 [NOTICE] ozmod_zt.c:1182 Using DAHDI control device
2010-03-17 19:11:32.507233 [INFO] zap_io.c:2849 Loading IO from
/opt/freeswitch/mod/ozmod_zt.so [zt]
2010-03-17 19:11:32.507233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/zt.conf.
2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000
2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000
2010-03-17 19:11:32.508233 [INFO] zap_io.c:2595 auto-loaded 'zt'
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2616 created span 1 (FXS1)
of type zt
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [name]=[Brian May]
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [number]=[1002]
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [fxs-channel]=[1]
2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2671 setting trunk type to
'FXS' start(KEWL)
2010-03-17 19:11:32.509233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 1 fd 64 (No such device or address)
2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2616 created span 2 (FXS2)
of type zt
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [name]=[Brian May]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [number]=[1002]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [fxs-channel]=[2]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2671 setting trunk type to
'FXS' start(KEWL)
2010-03-17 19:11:32.511233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 2 fd 64 (No such device or address)
2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2616 created span 3 (FXO1)
of type zt
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [name]=[OpenZAP]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [number]=[0397551926]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [fxo-channel]=[3]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2658 setting trunk type to
'FXO' start(KEWL)
2010-03-17 19:11:32.512233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 3 fd 64 (No such device or address)
2010-03-17 19:11:32.512233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.513233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2616 created span 4 (FXO2)
of type zt
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [name]=[OpenZAP]
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [number]=[0397552345]
2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2629 span 4 [fxo-channel]=[4]
2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2658 setting trunk type to
'FXO' start(KEWL)
2010-03-17 19:11:32.514233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 4 fd 64 (No such device or address)
2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.515233 [INFO] zap_io.c:2773 Configured 0 channel(s)
2010-03-17 19:11:32.516233 [ERR] zap_io.c:3056 No modules configured!
2010-03-17 19:11:32.516233 [ERR] mod_openzap.c:3171 Error loading OpenZAP
2010-03-17 19:11:32.516233 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_openzap.so
**Module load routine returned an error**


Is there anything else I need to change when going from zaptel to dahdi?

The kernel modules seem to load ok, and /dev/dahdi/* looks good too.
-- 
Brian May <brian at microcomaustralia.com.au>


From irmatov at gmail.com  Wed Mar 17 01:32:50 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Wed, 17 Mar 2010 13:32:50 +0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
Message-ID: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>

On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org> wrote:
> It already works use the respond application in the dialplan to respond 484

Thanks for the hint, Brian. I have a SoftX3000 softswitch sending
calls to FreeSWITCH. SoftX3000 is configured to send anything between
5 and 20 digits. I want FreeSWITCH to respond 484 until it receives
enough digits to make a call.

Here is an excerpt from my dialplan:

<extension name="public_extensions">
  <condition field="destination_number" expression="^006[0-9]{0,13}$">
    <action application="respond" data="484"/>
  </condition>
</extension>

In the first INVITE, softswitch sends 5 digits: 00610. FreeSWITCH
answers 484 just as I want it to. Softswitch sends another INVITE,
this time with 6 digits: 006107, but *without* SDP. Missing SDP upsets
FreeSWITCH and it responds 480, call finishes. Here is log from
FreeSWITCH:

2010-03-17 09:45:07.329047 [NOTICE] switch_channel.c:669 New Channel
sofia/external/1220139 at 10.0.2.5 [dc6f5d24-317f-11df-9775-c1f542b963a5]
2010-03-17 09:45:07.331047 [INFO] mod_dialplan_xml.c:418 Processing
1220139->00610 in context public
2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:185
sofia/external/1220139 at 10.0.2.5 has executed the last dialplan
instruction, hanging up.
2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:187
Hangup sofia/external/1220139 at 10.0.2.5 [CS_EXECUTE] [NORMAL_CLEARING]
2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1179 Session
3 (sofia/external/1220139 at 10.0.2.5) Ended
2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
2010-03-17 09:45:08.448839 [NOTICE] switch_channel.c:669 New Channel
sofia/external/1220139 at 10.0.2.5 [dd1a3776-317f-11df-9776-c1f542b963a5]
2010-03-17 09:45:08.448839 [INFO] sofia.c:4388 No SDP in INVITE and
3pcc not enabled, hanging up.
2010-03-17 09:45:08.449840 [NOTICE] sofia.c:4390 Hangup
sofia/external/1220139 at 10.0.2.5 [CS_NEW] [MANDATORY_IE_MISSING]
2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1179 Session
4 (sofia/external/1220139 at 10.0.2.5) Ended
2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]


And here are SIP messages, capture made with ngrep:

U 10.0.2.5:5070 -> 172.16.12.11:5090
INVITE sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
Supported: 100rel.
P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
Content-Length: 263.
Content-Type: application/sdp.
.
v=0.
o=HuaweiSoftX3000 98044 98044 IN IP4 10.0.2.5.
s=Sip Call.
c=IN IP4 172.16.12.1.
t=0 0.
m=audio 50000 RTP/AVP 8 0 18 4 2.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=fmtp:18 annexb=yes.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
CSeq: 1 INVITE.
Contact: <sip:00610 at 172.16.12.11:5090;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.	
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Length: 0.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
ACK sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
CSeq: 1 ACK.
Max-Forwards: 70.
Content-Length: 0.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
INVITE sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING".
Content-Length: 0.
P-Asserted-Identity: "006107" <sip:006107 at 172.16.12.11>.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
ACK sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
CSeq: 1 ACK.
Max-Forwards: 70.
Content-Length: 0.

RFC3578 says:

"The new INVITE has the same Call-ID and the same From header field
including the tag as the first INVITE sent, but has an updated
Request-URI."

SoftX3000 sends second INVITE with different Call-ID. Is there
anything I can do on FreeSWITCH to solve this problem and continue
receiving additional digits?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From dujinfang at gmail.com  Wed Mar 17 03:01:43 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 17 Mar 2010 18:01:43 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
Message-ID: <23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> Are you bridging to sip or some ivr and making dingaling wait until the call
> answers?
>

no I was originate dingaling/... &echo,   also tried originate
dingaling/...  extension XML public (which will hit dialplan and
answer() after client answers), neither work.


> gtalk is sort of missing key telephony features like the concept of pdd and
> indication progress etc.
>
> Try answering the call in your dialplan before you bridge.
>
> I think the gtalk client is getting impatient because you did not answer
> soon enough with an ip choice
> so it starts sending all these extra ones at you and gave up on the one we
> chose.
>
>

I tested on another server runing FreeSWITCH Version 1.0.trunk
(16958M), it works.

see http://pastebin.freeswitch.org/12463

interesting, messages like the same, but I see the working version has
one more line on line 174:

http://pastebin.freeswitch.org/12463 line: 173 to 175
2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
[dingaling/gmail.com/seven at idapted.com] has been answered
2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
(dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
CS_ROUTING


http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)

2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
(dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
CS_ROUTING

Do you see any other difference? Thanks.


> On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > FS would prefer you run your kernel at 1000hz you are probably on a VM
>> > if
>> > its 10000hz
>> >
>>
>> It's a ?physical machine. And I'm also running 20 skype clients. as
>> 100HZ is recommended for skypiax. I think the kernel should be 100 or
>> 250 HZ. weird.
>>
>>
>> here is the dingaling log with dl_debug on
>>
>> http://pastebin.freeswitch.org/12453
>> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>>
>>
>> still get this message on client:
>>
>> Sorry! The voice chat with language_lesson failed because of a network
>> problem at 5:39 PM. Please try again.
>> ?Click here to report this to Google
>>
>>
>>
>> Thanks.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> >
>> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> >> Windows client and web site on Mac. They both work on the other two
>> >> instances of FreeSWITCH I mentioned in this thread.
>> >>
>> >> btw, can you give me a hint on the timer warning?
>> >>
>> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> > you would have to execute
>> >> >
>> >> > dl_debug on
>> >> >
>> >> > to get a trace that would tell us anything about that?
>> >> >
>> >> > Are you using the windows gtalk client or the web site?
>> >> > We are following google who have been pushing for the online client
>> >> > embedded
>> >> > in the gmail website.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> here is the client:
>> >> >>
>> >> >> <profile type="client">
>> >> >> ?<param name="name" value="gmail.com"/>
>> >> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> >> ?<param name="password" value="password"/>
>> >> >> ?<param name="server" value="talk.google.com"/>
>> >> >> ?<param name="dialplan" value="XML"/>
>> >> >> ?<param name="contex" value="public"/>
>> >> >> ?<param name="message" value="FreeSWITCH"/>
>> >> >> ?<param name="rtp-ip" value="auto"/>
>> >> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> >> ?<param name="auto-login" value="true"/>
>> >> >> ?<param name="auto-reply" value="Welcome"/>
>> >> >> ?<param name="sasl" value="plain"/>
>> >> >> ?<param name="tls" value="true"/>
>> >> >> ?<param name="exten" value="5000"/>
>> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> >> </profile>
>> >> >>
>> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> >> > ALL,
>> >> >> >
>> >> >> > I just upgrade FS to 16992.
>> >> >> >
>> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
>> >> >> > sounds
>> >> >> > ok.
>> >> >> >
>> >> >> >
>> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> >> >> > detected!
>> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> >> > experience audio problems.
>> >> >> >
>> >> >> > uname -a
>> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05
>> >> >> > UTC
>> >> >> > 2008 i686 GNU/Linux
>> >> >> >
>> >> >> >
>> >> >> > I used a script to check the interupts
>> >> >> >
>> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >> >
>> >> >> >
>> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
>> >> >> > answer.
>> >> >> > I
>> >> >> > originate a call from FS, the console blocks, gtalk client ring
>> >> >> > and
>> >> >> > it
>> >> >> > looks ok when I hit answer, but no sound, FS console still blocks.
>> >> >> > a
>> >> >> > few seconds later gtalk client shows:
>> >> >> >
>> >> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> >> > network
>> >> >> > problem at 5:50 PM. Please try again.
>> >> >> > ?Click here to report this to Google.
>> >> >> >
>> >> >> >
>> >> >> > here is a log with bgapi:
>> >> >> > http://pastebin.freeswitch.org/12434
>> >> >> >
>> >> >> >
>> >> >> > rev 14696 works on the same server same conf. another server
>> >> >> > running
>> >> >> > 16958M works at the same time. I have no clue to check, is it
>> >> >> > related
>> >> >> > to the timer as in 1)? can some help me to take a look? let me
>> >> >> > know
>> >> >> > if
>> >> >> > you need "dl_debug on" logs.
>> >> >> >
>> >> >> > Thanks.
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ivdreg at gmail.com  Wed Mar 17 03:42:20 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 12:42:20 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
Message-ID: <b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>

Thanks again Rupa,

Is it possible to have more than one limit per call ? I have an issue in
this case with calls per period limiting.

This is part of my dial-plan before bridge:

<action application="limit_hash" data="caller ${destination_number} 5
!INCOMING_CALL_BARRED" />
<action application="limit_hash" data="crcd_${destination_number}
${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
<action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />

These are vars for 3 calls (not more than 10 sec between them):

variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]
--
variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]
--
variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]

I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
case. Is it true ? Does limit_usage and limit_rate that have no realm need
to exist also ?

Thanks

2010/3/16 Rupa Schomaker <rupa at rupa.com>

> limit and limit_hash should work the same with regards to disconnect cause.
>  If you see a difference, let me know.
>
>
> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Thanks Rupa,
>>
>> This was good clarification. I want to use mod_limit via 4 servers but
>> because traffic is equally balanced between them I will use limit_hash and
>> calculate how many calls every server will serve.
>>
>> One more question ? Is it necessary limit to transfer to limit_exceeded or
>> there is a way to set directly disconnect cause like in limit_hash ?
>>
>> Cheers
>>
>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>
>> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
>>> to a database when using limit_hash.  If you want to use the database, then
>>> use plain limit.
>>>
>>> 2) If you want a max calls in a 24hrs period use you rate with
>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>
>>> limit_hash realm id 50/86400
>>>
>>> should do what you want -- from memory.  double check the wiki/help to
>>> verify syntax.
>>>
>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>
>>>> Hi All,
>>>>
>>>> I have some questions regarding mod_limit.
>>>>
>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>
>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>
>>>> and they are applied on inbound call profile in same extension.
>>>>
>>>>
>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>> call but not calls processed as count. Any idea ?
>>>>
>>>> Thanks for any help
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Wed Mar 17 03:57:52 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 17 Mar 2010 05:57:52 -0500
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
References: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
Message-ID: <4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>

Brian May wrote:

> Is there anything else I need to change when going from zaptel to dahdi?
> 
> The kernel modules seem to load ok, and /dev/dahdi/* looks good too.

Make the appropriate changes to the configuration files in /etc/dahdi/. Take a look at the files in /proc/dahdi/.

--
Russell Mosemann



From ivdreg at gmail.com  Wed Mar 17 04:03:42 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 13:03:42 +0200
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
References: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
Message-ID: <b437a0e1003170403yc389c99q433156994fb51c46@mail.gmail.com>

Hi this is a little example from my perl code that do that:

array random_be contains randomized IP addresses of next hops.
there also hash:
my %hunt_error=(
        NO_USER_RESPONSE => 1,
        NETWORK_OUT_OF_ORDER => 1,
        DESTINATION_OUT_OF_ORDER => 1,
        SYSTEM_SHUTDOWN => 1,
);

If error is defined in this array we hunt else drop call. I assume that
these errors are internally generated from this FS instance or comes first
remote hop. So I need to hunt over next hop. But if error comes from last
server I drop call and do not hunt over rest of my SIP servers.

for (my $i=0;$i <= scalar(@random_be) ; $i++) {
        my $dst_gw = shift(@random_be);
        print "GW:[$dst_gw]\n";
        $bridge = "sofia/internal_out/$called\@$dst_gw";
        $session->execute('bridge',"$bridge");
        $berror = $session->getVariable("originate_disposition");
        if (defined($berror)){
                fprint("\tBridge Error - Check continue or not\n");
                if (!defined($hunt_error{$berror})){
                        fprint("\t HANGUP - ERROR NOT DEFINED");

$session->execute('export','continue_on_fail=false');
                        $session->hangup();
                        return (1);
                }
        } else { last; } ;  ### NO originate_disposition so successful
bridge
}

Suppose this will help. Not in Lua but idea is the same.

Cheers


2010/3/14 rentmycoder rentmycoder <rentmycoder at gmail.com>

> > Pass them as arguments to your luarun hangup hook.
> And what about variables are not available before the call setup (like
> uuid), only during the call progress or after the hangup???
> I would need to access the originate_disposition variable...
>
> >> 'luarun hangup.lua ${originate_disposition}'
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> any other hint???
>
>
> > ---------- Forwarded message ----------
> > From: Brian West <brian at freeswitch.org>
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Sun, 14 Mar 2010 13:00:10 -0500
> > Subject: Re: [Freeswitch-users] how to access session variables after
> hangup???
> > Pass them as arguments to your luarun hangup hook.
> >
> > /b
> >
> > On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:
> >
> >> Hi guys,
> >> Freeswitch rocks!
> >> except the documentation:(
> >> I'm not able to figure out how to access session variables after
> hangup...
> >> I try to dial out on a gateway and detect it the called party hangs up
> >> before bridge...
> >>
> >> bgapi originate
> >>
> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
> >> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
> >>
> >> I cannot access the variables using:
> >> 1. script arguments:
> >> 'luarun hangup.lua ${uuid}'}
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> >> (${api_hangup_hook} contains a variable)
> >>
> >> 2. session:getVariable("originate_disposition")
> >> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
> >> to call global 'getVar' (a nil value)
> >>
> >> 3. params:getHeader("variable_sip_req_uri")
> >> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
> >> to index global 'params' (a nil value)
> >>
> >> I have asked this before on the dev list too, but Anthony didn't gave
> >> me a clean answer here:
> >>
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
> >>
> >> "In the script you can access the data from the env event which is
> >> automatically created for you and it
> >> contains all the channel variables from that departed channel."
> >>
> >> Ok, but how?????
> >>
> >> Please help...
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Wed Mar 17 05:01:15 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 09:01:15 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003170049.06565.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
	<201003170049.06565.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>

Rob, I believe that the core has failed to load. I have not added any type
of check on the code level for those cases yet as I believed it would never
really fail to load. Those cases have been more frequent than what I
originally expected so I will add that to code today (hopefully later
tonight).

Meanwhile, what you can do is check if the ports are open or not on the OS.
I am not sure how to do that on Windows. Or, maybe we already have logs on
the log folder?

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> Is there something wrong with my build?
>
> fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> [Socket
> Connection Error]" with fscomm running.
>
> RD
>
> On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > With FSComm running you can use fs_cli to try pa list. If u can't list
> the
> > devices that would be the problem. I'm going to try to dig into this
> > further on 1 of my desktops w/ pulseaudio.
> >
> > -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xengelpublicx at gmail.com  Wed Mar 17 05:22:11 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Wed, 17 Mar 2010 15:22:11 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
Message-ID: <eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>

Yes

On Wed, Mar 17, 2010 at 12:19 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Mar 15, 2010 at 1:21 PM, Vladimir Elizarov <xengelpublicx at gmail.com>
> wrote:
>>
>> Hello.
>>
>> How can I intercept a fifo? I tried to use the example of intecept
>> default dialplan, but in no fifo dialed_extension.
>
> Are you trying to intercept the call that is ringing at a FIFO agent's
> phone?
> -MC
>
>>
>> ? ?<extension name="operators">
>> ? ? ?<condition field="destination_number" expression="^operators$">
>> ? ? ? ? ? ?<action application="bind_meta_app" data="2 b s
>>
>> record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>> ? ? ? ? ? ?<action application="hash"
>>
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>> ? ? ? ? ? ?<action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>> ? ? ? ? ? ?<action application="set"
>>
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>> ? ? ? ? ? ?<action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>> ? ? ? ? ? ?<action application="answer"/>
>> ? ? ? ? ? ?<action application="playback"
>> data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
>> ? ? ? ? ? ?<action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>> ? ? ? ? ? ?<action application="set" data="fifo_music=$${hold_music}"/>
>> ? ? ? ? ? ?<action application="fifo" data="operators@$${domain} in
>> undef"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> ? ?<fifo name="operators@$${domain}" importance="0">
>> ? ? ?<member timeout="15" simo="1"
>>
>> lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member>
>> ? ?</fifo>
>> ?</fifos>
>> --
>> Best regards, Vladimir Elizarov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov


From brian at freeswitch.org  Wed Mar 17 06:18:54 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:18:54 -0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
Message-ID: <A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>


On Mar 17, 2010, at 3:32 AM, Timur Irmatov wrote:

> On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org> wrote:
>> It already works use the respond application in the dialplan to respond 484
> 
> 
> RFC3578 says:
> 
> "The new INVITE has the same Call-ID and the same From header field
> including the tag as the first INVITE sent, but has an updated
> Request-URI."
> 
> SoftX3000 sends second INVITE with different Call-ID. Is there
> anything I can do on FreeSWITCH to solve this problem and continue
> receiving additional digits?

NO, Why should we even try?  You paid Huawei for this switch that is doing something wrong.  Contact them and make them fix it... if they won't then you have to ask yourself why they should?  You paid them... they have your money... they don't care.

/b



From brian at freeswitch.org  Wed Mar 17 06:19:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:19:15 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268811580808-4748784.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
Message-ID: <E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>

First off are you on SVN Trunk?

/b

On Mar 17, 2010, at 2:39 AM, Maciej Aniserowicz wrote:

> Hi again,
> I don't quite understand why both of them would not be parked... in logs
> (from http://pastebin.freeswitch.org/12432) I can see this:

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From brian at freeswitch.org  Wed Mar 17 06:23:03 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:23:03 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
Message-ID: <EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>

This is where it gets hairy... Yes the MPL says you have to share your modifications with the original author but as far as I can tell you don't have to distribute to cause this clause to go into effect.  

Its like this:

You change switch_ivr_originate.c, you HAVE to share those changes with Anthony.
You create your own mod_kewlshit.c, its yours to do with what you want and you don't have to share.

You change any code that is included with the download YOU MUST share with the original author.

/b


On Mar 16, 2010, at 11:30 PM, jay binks wrote:

> Im not on the FS dev team, but I think you may be a little mistaken.
> 
> generally your free to make modifications to open source code ( for
> internal use )
> however its when you distribute the resultant product, then you have
> to make the source available..
> 
> if your only using it in-house, I dont believe your required to
> contribute the code back.
> ( however your going to end up having issues maintaining your patch
> set over a long period.. like can be seen here )
> 
> see here :
> 
>> must be made available in Source Code form under the terms of this License either on the
>> same media as an Executable version
> 
> if your not distributing your executable to any 3rd parties... then I
> dont believe you have to distribute the modified source..
> 
> Correct me if Im wrong though guys...
> 
> Jay
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From rupa at rupa.com  Wed Mar 17 06:41:08 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 17 Mar 2010 08:41:08 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
Message-ID: <b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>

That should work.  Can you open a jira against mod_limit and assign it to
me?  Thanks.

On Wed, Mar 17, 2010 at 5:42 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Thanks again Rupa,
>
> Is it possible to have more than one limit per call ? I have an issue in
> this case with calls per period limiting.
>
> This is part of my dial-plan before bridge:
>
> <action application="limit_hash" data="caller ${destination_number} 5
> !INCOMING_CALL_BARRED" />
> <action application="limit_hash" data="crcd_${destination_number}
> ${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
> <action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
> ${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />
>
> These are vars for 3 calls (not more than 10 sec between them):
>
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
> --
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
> --
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
>
> I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
> case. Is it true ? Does limit_usage and limit_rate that have no realm need
> to exist also ?
>
> Thanks
>
>
> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>
>> limit and limit_hash should work the same with regards to disconnect
>> cause.  If you see a difference, let me know.
>>
>>
>> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Thanks Rupa,
>>>
>>> This was good clarification. I want to use mod_limit via 4 servers but
>>> because traffic is equally balanced between them I will use limit_hash and
>>> calculate how many calls every server will serve.
>>>
>>> One more question ? Is it necessary limit to transfer to limit_exceeded
>>> or there is a way to set directly disconnect cause like in limit_hash ?
>>>
>>> Cheers
>>>
>>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>>
>>> 1) limit_hash use an in-memory hash for it's storage. There is nor
>>>> storing to a database when using limit_hash.  If you want to use the
>>>> database, then use plain limit.
>>>>
>>>> 2) If you want a max calls in a 24hrs period use you rate with
>>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>>
>>>> limit_hash realm id 50/86400
>>>>
>>>> should do what you want -- from memory.  double check the wiki/help to
>>>> verify syntax.
>>>>
>>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>>
>>>>> Hi All,
>>>>>
>>>>> I have some questions regarding mod_limit.
>>>>>
>>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>>
>>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>>
>>>>> and they are applied on inbound call profile in same extension.
>>>>>
>>>>>
>>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>>> call but not calls processed as count. Any idea ?
>>>>>
>>>>> Thanks for any help
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From delorenzodesign at gmail.com  Wed Mar 17 07:00:56 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 17 Mar 2010 10:00:56 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
Message-ID: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>

How do I do that in Lua?  session:execute("dtmf","start") doesn't work.

On Tue, Mar 16, 2010 at 11:38 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
> -MC
>
> Sent from my iPhone
>
> On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <delorenzodesign at gmail.com>
> wrote:
>
> So I'm making calls using VoiceNetwork, are you suggesting there's a
> problem there?  Or on the line where the call is received (I've tried on a
> couple of different networks/providers)?
>
> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
> lloyd.aloysius at gmail.com> wrote:
>
>> You inbound DID Provider
>>
>>
>>
>> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>> delorenzodesign at gmail.com> wrote:
>>
>>> What do you mean about the provider?  I'm not sure I understand that
>>> part.  I'm trying to call a number and have the call's recipient type the
>>> digits in.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> here is the test script I used
>>>>
>>>> session:answer();
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> "enter-digits.wav", "", "\\d+");
>>>>
>>>> Here is the console log
>>>>
>>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>>> playing file
>>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 1:800*
>>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 2:640*
>>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 3:800*
>>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 4:640*
>>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>>>> [1234][\d+]
>>>>
>>>> You may have some issue with the provider. Check using a local
>>>> extension.
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> Ok, I updated the console log level, and there are no DTMF values being
>>>>> detected:
>>>>>
>>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>>> [soft] 160 bytes per 20ms
>>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>>> sofia/external/19736327407!
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>> sofia/external/19736327407 entering state [completing][200]
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>>> v=0
>>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>>> s=session
>>>>> c=IN IP4 74.51.38.156
>>>>> t=0 0
>>>>> m=audio 15150 RTP/AVP 0 101
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>> sofia/external/19736327407 entering state [ready][200]
>>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>>> [sofia/external/19736327407] has been answered
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107
>>>>> Originate Resulted in Success: [sofia/external/19736327407]
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>>> CS_SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/external/19736327407 [BREAK]
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>>> Activated L16 at 8000hz 1 channels 20ms
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>>> confirmed.
>>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>>> playing file
>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>>> during call from playAndGetDigits:
>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>>> out? no
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>>> lloyd.aloysius at gmail.com> wrote:
>>>>>
>>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>>> console log , while you are pressing the keys.
>>>>>>
>>>>>> Lloyd
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>
>>>>>>> So I tried Michael's suggestion of:
>>>>>>>
>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>
>>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>>> being detected?
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <<msc at freeswitch.org>
>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>>
>>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>>> possible.
>>>>>>>>>
>>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>     session =
>>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>>> "/1" .. number_to_call);
>>>>>>>>>
>>>>>>>>>     while(session:ready()) do
>>>>>>>>>        session:answer();
>>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>>
>>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>>
>>>>>>>>
>>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>>
>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>
>>>>>>>> Let us know if that works. Oh, one other thing: you might want to
>>>>>>>> add an invalid message because playAndGetDigits will handle invalid input
>>>>>>>> for you.
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>>
>>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>>        if(check_success) then
>>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>>> received from this check.\n");
>>>>>>>>>           break;
>>>>>>>>>        else
>>>>>>>>>           -- not a positive response
>>>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>>             freeswitch.consoleLog("info", "We've reached our
>>>>>>>>> maximum attempts for this number.\n");
>>>>>>>>>             break;
>>>>>>>>>           end
>>>>>>>>>        end
>>>>>>>>>     end
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>
>>
> _______________________________________________
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>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From anthony.minessale at gmail.com  Wed Mar 17 07:14:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 09:14:56 -0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
	<A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>
Message-ID: <191c3a031003170714s24a14ce0oa58ec91bc0ec84a@mail.gmail.com>

Not to mention we would have to hack the sofia sip stack in ways I can't
begin to imagine.
It so horribly violates the spec its a lost cause.


On Wed, Mar 17, 2010 at 8:18 AM, Brian West <brian at freeswitch.org> wrote:

>
> On Mar 17, 2010, at 3:32 AM, Timur Irmatov wrote:
>
> > On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> It already works use the respond application in the dialplan to respond
> 484
> >
> >
> > RFC3578 says:
> >
> > "The new INVITE has the same Call-ID and the same From header field
> > including the tag as the first INVITE sent, but has an updated
> > Request-URI."
> >
> > SoftX3000 sends second INVITE with different Call-ID. Is there
> > anything I can do on FreeSWITCH to solve this problem and continue
> > receiving additional digits?
>
> NO, Why should we even try?  You paid Huawei for this switch that is doing
> something wrong.  Contact them and make them fix it... if they won't then
> you have to ask yourself why they should?  You paid them... they have your
> money... they don't care.
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 07:16:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 09:16:44 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
Message-ID: <191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>

Were you using gtalk and FS both behind nat with the public gtalk server by
any chance?
This setup requires special configuration.


On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > Are you bridging to sip or some ivr and making dingaling wait until the
> call
> > answers?
> >
>
> no I was originate dingaling/... &echo,   also tried originate
> dingaling/...  extension XML public (which will hit dialplan and
> answer() after client answers), neither work.
>
>
> > gtalk is sort of missing key telephony features like the concept of pdd
> and
> > indication progress etc.
> >
> > Try answering the call in your dialplan before you bridge.
> >
> > I think the gtalk client is getting impatient because you did not answer
> > soon enough with an ip choice
> > so it starts sending all these extra ones at you and gave up on the one
> we
> > chose.
> >
> >
>
> I tested on another server runing FreeSWITCH Version 1.0.trunk
> (16958M), it works.
>
> see http://pastebin.freeswitch.org/12463
>
> interesting, messages like the same, but I see the working version has
> one more line on line 174:
>
> http://pastebin.freeswitch.org/12463 line: 173 to 175
> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
> [dingaling/gmail.com/seven at idapted.com] has been answered
> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> CS_ROUTING
>
>
> http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)
>
> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> CS_ROUTING
>
> Do you see any other difference? Thanks.
>
>
> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > FS would prefer you run your kernel at 1000hz you are probably on a VM
> >> > if
> >> > its 10000hz
> >> >
> >>
> >> It's a  physical machine. And I'm also running 20 skype clients. as
> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> >> 250 HZ. weird.
> >>
> >>
> >> here is the dingaling log with dl_debug on
> >>
> >> http://pastebin.freeswitch.org/12453
> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
> >>
> >>
> >> still get this message on client:
> >>
> >> Sorry! The voice chat with language_lesson failed because of a network
> >> problem at 5:39 PM. Please try again.
> >>  Click here to report this to Google
> >>
> >>
> >>
> >> Thanks.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> >
> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> >> Windows client and web site on Mac. They both work on the other two
> >> >> instances of FreeSWITCH I mentioned in this thread.
> >> >>
> >> >> btw, can you give me a hint on the timer warning?
> >> >>
> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> > you would have to execute
> >> >> >
> >> >> > dl_debug on
> >> >> >
> >> >> > to get a trace that would tell us anything about that?
> >> >> >
> >> >> > Are you using the windows gtalk client or the web site?
> >> >> > We are following google who have been pushing for the online client
> >> >> > embedded
> >> >> > in the gmail website.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> here is the client:
> >> >> >>
> >> >> >> <profile type="client">
> >> >> >>  <param name="name" value="gmail.com"/>
> >> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >> >>  <param name="password" value="password"/>
> >> >> >>  <param name="server" value="talk.google.com"/>
> >> >> >>  <param name="dialplan" value="XML"/>
> >> >> >>  <param name="contex" value="public"/>
> >> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >> >>  <param name="rtp-ip" value="auto"/>
> >> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >> >>  <param name="auto-login" value="true"/>
> >> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >> >>  <param name="sasl" value="plain"/>
> >> >> >>  <param name="tls" value="true"/>
> >> >> >>  <param name="exten" value="5000"/>
> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> >> >> </profile>
> >> >> >>
> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> >> > ALL,
> >> >> >> >
> >> >> >> > I just upgrade FS to 16992.
> >> >> >> >
> >> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
> >> >> >> > sounds
> >> >> >> > ok.
> >> >> >> >
> >> >> >> >
> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
> microseconds
> >> >> >> > detected!
> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> >> >> > experience audio problems.
> >> >> >> >
> >> >> >> > uname -a
> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
> 08:27:05
> >> >> >> > UTC
> >> >> >> > 2008 i686 GNU/Linux
> >> >> >> >
> >> >> >> >
> >> >> >> > I used a script to check the interupts
> >> >> >> >
> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >> >
> >> >> >> >
> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> >> >> >> > answer.
> >> >> >> > I
> >> >> >> > originate a call from FS, the console blocks, gtalk client ring
> >> >> >> > and
> >> >> >> > it
> >> >> >> > looks ok when I hit answer, but no sound, FS console still
> blocks.
> >> >> >> > a
> >> >> >> > few seconds later gtalk client shows:
> >> >> >> >
> >> >> >> > Sorry! The voice chat with language_lesson failed because of a
> >> >> >> > network
> >> >> >> > problem at 5:50 PM. Please try again.
> >> >> >> >  Click here to report this to Google.
> >> >> >> >
> >> >> >> >
> >> >> >> > here is a log with bgapi:
> >> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >> >
> >> >> >> >
> >> >> >> > rev 14696 works on the same server same conf. another server
> >> >> >> > running
> >> >> >> > 16958M works at the same time. I have no clue to check, is it
> >> >> >> > related
> >> >> >> > to the timer as in 1)? can some help me to take a look? let me
> >> >> >> > know
> >> >> >> > if
> >> >> >> > you need "dl_debug on" logs.
> >> >> >> >
> >> >> >> > Thanks.
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ivdreg at gmail.com  Wed Mar 17 07:33:49 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 16:33:49 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
	<b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
Message-ID: <b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>

Jira opened. It's for my first time so may be not perfect.

2010/3/17 Rupa Schomaker <rupa at rupa.com>

> That should work.  Can you open a jira against mod_limit and assign it to
> me?  Thanks.
>
>
> On Wed, Mar 17, 2010 at 5:42 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Thanks again Rupa,
>>
>> Is it possible to have more than one limit per call ? I have an issue in
>> this case with calls per period limiting.
>>
>> This is part of my dial-plan before bridge:
>>
>> <action application="limit_hash" data="caller ${destination_number} 5
>> !INCOMING_CALL_BARRED" />
>> <action application="limit_hash" data="crcd_${destination_number}
>> ${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
>> <action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
>> ${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />
>>
>> These are vars for 3 calls (not more than 10 sec between them):
>>
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>> --
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>> --
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>>
>> I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
>> case. Is it true ? Does limit_usage and limit_rate that have no realm need
>> to exist also ?
>>
>> Thanks
>>
>>
>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>
>>> limit and limit_hash should work the same with regards to disconnect
>>> cause.  If you see a difference, let me know.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>>
>>>> Thanks Rupa,
>>>>
>>>> This was good clarification. I want to use mod_limit via 4 servers but
>>>> because traffic is equally balanced between them I will use limit_hash and
>>>> calculate how many calls every server will serve.
>>>>
>>>> One more question ? Is it necessary limit to transfer to limit_exceeded
>>>> or there is a way to set directly disconnect cause like in limit_hash ?
>>>>
>>>> Cheers
>>>>
>>>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>>>
>>>> 1) limit_hash use an in-memory hash for it's storage. There is nor
>>>>> storing to a database when using limit_hash.  If you want to use the
>>>>> database, then use plain limit.
>>>>>
>>>>> 2) If you want a max calls in a 24hrs period use you rate with
>>>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>>>
>>>>> limit_hash realm id 50/86400
>>>>>
>>>>> should do what you want -- from memory.  double check the wiki/help to
>>>>> verify syntax.
>>>>>
>>>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>>>
>>>>>> Hi All,
>>>>>>
>>>>>> I have some questions regarding mod_limit.
>>>>>>
>>>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>>>
>>>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>>>
>>>>>> and they are applied on inbound call profile in same extension.
>>>>>>
>>>>>>
>>>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>>>> call but not calls processed as count. Any idea ?
>>>>>>
>>>>>> Thanks for any help
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 07:41:05 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Mar 2010 07:41:05 -0700
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
Message-ID: <147345D8-6862-4B71-AF6A-27D0C686A78D@freeswitch.org>

Does the Huawei even claim to support rfc3578? If so you have proof  
that they are not following the spec. Contact them to find out what's  
going on. I'd love to see what happens if you ask them for a refund...
-MC

Sent from my iPhone

On Mar 17, 2010, at 1:32 AM, Timur Irmatov <irmatov at gmail.com> wrote:

> On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> It already works use the respond application in the dialplan to  
>> respond 484
>
> Thanks for the hint, Brian. I have a SoftX3000 softswitch sending
> calls to FreeSWITCH. SoftX3000 is configured to send anything between
> 5 and 20 digits. I want FreeSWITCH to respond 484 until it receives
> enough digits to make a call.
>
> Here is an excerpt from my dialplan:
>
> <extension name="public_extensions">
>  <condition field="destination_number" expression="^006[0-9]{0,13}$">
>    <action application="respond" data="484"/>
>  </condition>
> </extension>
>
> In the first INVITE, softswitch sends 5 digits: 00610. FreeSWITCH
> answers 484 just as I want it to. Softswitch sends another INVITE,
> this time with 6 digits: 006107, but *without* SDP. Missing SDP upsets
> FreeSWITCH and it responds 480, call finishes. Here is log from
> FreeSWITCH:
>
> 2010-03-17 09:45:07.329047 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/1220139 at 10.0.2.5 [dc6f5d24-317f-11df-9775-c1f542b963a5]
> 2010-03-17 09:45:07.331047 [INFO] mod_dialplan_xml.c:418 Processing
> 1220139->00610 in context public
> 2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:185
> sofia/external/1220139 at 10.0.2.5 has executed the last dialplan
> instruction, hanging up.
> 2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:187
> Hangup sofia/external/1220139 at 10.0.2.5 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1179 Session
> 3 (sofia/external/1220139 at 10.0.2.5) Ended
> 2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
> 2010-03-17 09:45:08.448839 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/1220139 at 10.0.2.5 [dd1a3776-317f-11df-9776-c1f542b963a5]
> 2010-03-17 09:45:08.448839 [INFO] sofia.c:4388 No SDP in INVITE and
> 3pcc not enabled, hanging up.
> 2010-03-17 09:45:08.449840 [NOTICE] sofia.c:4390 Hangup
> sofia/external/1220139 at 10.0.2.5 [CS_NEW] [MANDATORY_IE_MISSING]
> 2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1179 Session
> 4 (sofia/external/1220139 at 10.0.2.5) Ended
> 2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
>
>
> And here are SIP messages, capture made with ngrep:
>
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> INVITE sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>.
> CSeq: 1 INVITE.
> Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
> Supported: 100rel.
> P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
> User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
> Max-Forwards: 70.
> Allow:  
> INVITE, 
> ACK, 
> CANCEL, 
> OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
> Content-Length: 263.
> Content-Type: application/sdp.
> .
> v=0.
> o=HuaweiSoftX3000 98044 98044 IN IP4 10.0.2.5.
> s=Sip Call.
> c=IN IP4 172.16.12.1.
> t=0 0.
> m=audio 50000 RTP/AVP 8 0 18 4 2.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:2 G726-32/8000.
> a=fmtp:18 annexb=yes.
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 484 Address Incomplete.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> CSeq: 1 INVITE.
> Contact: <sip:00610 at 172.16.12.11:5090;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Length: 0.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> ACK sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
> CSeq: 1 ACK.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> INVITE sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>.
> CSeq: 1 INVITE.
> Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
> P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
> User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
> Max-Forwards: 70.
> Allow:  
> INVITE, 
> ACK, 
> CANCEL, 
> OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 480 Temporarily Unavailable.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING".
> Content-Length: 0.
> P-Asserted-Identity: "006107" <sip:006107 at 172.16.12.11>.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> ACK sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
> CSeq: 1 ACK.
> Max-Forwards: 70.
> Content-Length: 0.
>
> RFC3578 says:
>
> "The new INVITE has the same Call-ID and the same From header field
> including the tag as the first INVITE sent, but has an updated
> Request-URI."
>
> SoftX3000 sends second INVITE with different Call-ID. Is there
> anything I can do on FreeSWITCH to solve this problem and continue
> receiving additional digits?
>
>
> -- 
> Timur Irmatov, xmpp:irmatov at jabber.ru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From tculjaga at gmail.com  Wed Mar 17 07:46:57 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 17 Mar 2010 15:46:57 +0100
Subject: [Freeswitch-users] Release Cause in response message
Message-ID: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>

hello folks,


anyone knows how to pass a Release cause/ Disconnect Reason when firing
respond from a dialplan ... e.g.:

Ive tried with sip_h_Reason=Q.850;cause=27 but it is not working :)



   <extension name="parseResult">
      <condition field="destination_number" expression="^parseResult$"/>
      <condition field="${componentStatus}" expression="^1$"
break="on-true">
         <action application="log" data="INFO ######################## CALL
FAILED REASON 1' ##########\n"/>

         <action application="set" data="sip_h_Reason=Q.850;cause=16"/>

         <action application="respond" data="480 Temporarily Unavailable"/>
      </condition>
   </extension>


any hint?
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From delorenzodesign at gmail.com  Wed Mar 17 07:52:16 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 17 Mar 2010 10:52:16 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
	<5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
Message-ID: <5c9dcbfb1003170752rd35a064s9457006ab17630cb@mail.gmail.com>

Ok, so I updated my install to include OpenZap.  It looks like the keys are
being detected, but not being returned by playAndGetDigits.

2010-03-17 10:49:21.207160 [DEBUG] sofia.c:4152 Channel
sofia/external/19736327407 entering state [ready][200]
2010-03-17 10:49:21.208170 [NOTICE] sofia.c:4676 Channel
[sofia/external/19736327407] has been answered
2010-03-17 10:49:21.208170 [DEBUG] switch_ivr_originate.c:3140 Originate
Resulted in Success: [sofia/external/19736327407]
2010-03-17 10:49:21.208170 [DEBUG] switch_cpp.cpp:497
(sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
CS_SOFT_EXECUTE
2010-03-17 10:49:21.208170 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:21.209171 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:200
sofia/external/19736327407 Standard SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE going to sleep
2010-03-17 10:49:21.241184 [DEBUG] switch_rtp.c:2064 Correct ip/port
confirmed.
*2010-03-17 10:49:25.582703 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:1440*
2010-03-17 10:49:25.582703 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
*2010-03-17 10:49:27.342662 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:480*
*2010-03-17 10:49:28.042790 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:48*0
2010-03-17 10:49:31.829092 [NOTICE] sofia.c:481 Hangup
sofia/external/19736327407 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2010-03-17 10:49:31.829092 [DEBUG] switch_channel.c:2071 Send signal
sofia/external/19736327407 [KILL]
2010-03-17 10:49:31.829092 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.829092 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_HANGUP
2010-03-17 10:49:31.830229 [DEBUG] switch_core_state_machine.c:499
(sofia/external/19736327407) State HANGUP
2010-03-17 10:49:31.830229 [DEBUG] mod_sofia.c:405
sofia/external/19736327407 Overriding SIP cause 480 with 200 from the other
leg
2010-03-17 10:49:31.830229 [DEBUG] mod_sofia.c:411 Channel
sofia/external/19736327407 hanging up, cause: NORMAL_CLEARING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:46
sofia/external/19736327407 Standard HANGUP, cause: NORMAL_CLEARING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:499
(sofia/external/19736327407) State HANGUP going to sleep
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:333
(sofia/external/19736327407) State Change CS_HANGUP -> CS_REPORTING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_REPORTING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:590
(sofia/external/19736327407) State REPORTING
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:53
sofia/external/19736327407 Standard REPORTING, cause: NORMAL_CLEARING
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:590
(sofia/external/19736327407) State REPORTING going to sleep
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:327
(sofia/external/19736327407) State Change CS_REPORTING -> CS_DESTROY
2010-03-17 10:49:31.832242 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.832242 [DEBUG] switch_core_session.c:1161 Session 1
(sofia/external/19736327407) Locked, Waiting on external entities
*2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 Digits received
during call from playAndGetDigits:*
2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 Did this check out? no
2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 We need to try this
callee again.



On Wed, Mar 17, 2010 at 10:00 AM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> How do I do that in Lua?  session:execute("dtmf","start") doesn't work.
>
>
> On Tue, Mar 16, 2010 at 11:38 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>> So I'm making calls using VoiceNetwork, are you suggesting there's a
>> problem there?  Or on the line where the call is received (I've tried on a
>> couple of different networks/providers)?
>>
>> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> You inbound DID Provider
>>>
>>>
>>>
>>> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> What do you mean about the provider?  I'm not sure I understand that
>>>> part.  I'm trying to call a number and have the call's recipient type the
>>>> digits in.
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>> lloyd.aloysius at gmail.com> wrote:
>>>>
>>>>> here is the test script I used
>>>>>
>>>>> session:answer();
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> "enter-digits.wav", "", "\\d+");
>>>>>
>>>>> Here is the console log
>>>>>
>>>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>>>> playing file
>>>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 1:800*
>>>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 2:640*
>>>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 3:800*
>>>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 4:640*
>>>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test
>>>>> Regex [1234][\d+]
>>>>>
>>>>> You may have some issue with the provider. Check using a local
>>>>> extension.
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> Ok, I updated the console log level, and there are no DTMF values
>>>>>> being detected:
>>>>>>
>>>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>>>> [soft] 160 bytes per 20ms
>>>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>>>> sofia/external/19736327407!
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>>> sofia/external/19736327407 entering state [completing][200]
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>>>> v=0
>>>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>>>> s=session
>>>>>> c=IN IP4 74.51.38.156
>>>>>> t=0 0
>>>>>> m=audio 15150 RTP/AVP 0 101
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=silenceSupp:off - - - -
>>>>>> a=ptime:20
>>>>>>
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>>> sofia/external/19736327407 entering state [ready][200]
>>>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>>>> [sofia/external/19736327407] has been answered
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107
>>>>>> Originate Resulted in Success: [sofia/external/19736327407]
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>>>> signal sofia/external/19736327407 [BREAK]
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>>>> Activated L16 at 8000hz 1 channels 20ms
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>>>> confirmed.
>>>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>>>> playing file
>>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>>>> during call from playAndGetDigits:
>>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>>>> out? no
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>>>> lloyd.aloysius at gmail.com> wrote:
>>>>>>
>>>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>>>> console log , while you are pressing the keys.
>>>>>>>
>>>>>>> Lloyd
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>
>>>>>>>> So I tried Michael's suggestion of:
>>>>>>>>
>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>
>>>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>>>> being detected?
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <<msc at freeswitch.org>
>>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>>>
>>>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>>>> possible.
>>>>>>>>>>
>>>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>     session =
>>>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>>>> "/1" .. number_to_call);
>>>>>>>>>>
>>>>>>>>>>     while(session:ready()) do
>>>>>>>>>>        session:answer();
>>>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>>>
>>>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>>>
>>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>>
>>>>>>>>> Let us know if that works. Oh, one other thing: you might want to
>>>>>>>>> add an invalid message because playAndGetDigits will handle invalid input
>>>>>>>>> for you.
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>>>
>>>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>>>        if(check_success) then
>>>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>>>> received from this check.\n");
>>>>>>>>>>           break;
>>>>>>>>>>        else
>>>>>>>>>>           -- not a positive response
>>>>>>>>>>           -- if we reached our max attempts, we're finished move
>>>>>>>>>> on
>>>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>>>             freeswitch.consoleLog("info", "We've reached our
>>>>>>>>>> maximum attempts for this number.\n");
>>>>>>>>>>             break;
>>>>>>>>>>           end
>>>>>>>>>>        end
>>>>>>>>>>     end
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>
>
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From dujinfang at gmail.com  Wed Mar 17 08:03:12 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 17 Mar 2010 23:03:12 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
	<191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
Message-ID: <23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>

No.

FreeSWITCH runs on public IP.  my gtalk client behind NAT.  As I said,
old versions(e.g. 14696) work on the same server with the same config.

2010/3/17 Anthony Minessale <anthony.minessale at gmail.com>:
> Were you using gtalk and FS both behind nat with the public gtalk server by
> any chance?
> This setup requires special configuration.
>
>
> On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > Are you bridging to sip or some ivr and making dingaling wait until the
>> > call
>> > answers?
>> >
>>
>> no I was originate dingaling/... &echo, ? also tried originate
>> dingaling/... ?extension XML public (which will hit dialplan and
>> answer() after client answers), neither work.
>>
>>
>> > gtalk is sort of missing key telephony features like the concept of pdd
>> > and
>> > indication progress etc.
>> >
>> > Try answering the call in your dialplan before you bridge.
>> >
>> > I think the gtalk client is getting impatient because you did not answer
>> > soon enough with an ip choice
>> > so it starts sending all these extra ones at you and gave up on the one
>> > we
>> > chose.
>> >
>> >
>>
>> I tested on another server runing FreeSWITCH Version 1.0.trunk
>> (16958M), it works.
>>
>> see http://pastebin.freeswitch.org/12463
>>
>> interesting, messages like the same, but I see the working version has
>> one more line on line 174:
>>
>> http://pastebin.freeswitch.org/12463 line: 173 to 175
>> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
>> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
>> [dingaling/gmail.com/seven at idapted.com] has been answered
>> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
>> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
>> CS_ROUTING
>>
>>
>> http://pastebin.freeswitch.org/12453 ?line 217 - 218 ?(rev16992M)
>>
>> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
>> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
>> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
>> CS_ROUTING
>>
>> Do you see any other difference? Thanks.
>>
>>
>> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> > FS would prefer you run your kernel at 1000hz you are probably on a
>> >> > VM
>> >> > if
>> >> > its 10000hz
>> >> >
>> >>
>> >> It's a ?physical machine. And I'm also running 20 skype clients. as
>> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
>> >> 250 HZ. weird.
>> >>
>> >>
>> >> here is the dingaling log with dl_debug on
>> >>
>> >> http://pastebin.freeswitch.org/12453
>> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>> >>
>> >>
>> >> still get this message on client:
>> >>
>> >> Sorry! The voice chat with language_lesson failed because of a network
>> >> problem at 5:39 PM. Please try again.
>> >> ?Click here to report this to Google
>> >>
>> >>
>> >>
>> >> Thanks.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> >
>> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> >> >> Windows client and web site on Mac. They both work on the other two
>> >> >> instances of FreeSWITCH I mentioned in this thread.
>> >> >>
>> >> >> btw, can you give me a hint on the timer warning?
>> >> >>
>> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> >> > you would have to execute
>> >> >> >
>> >> >> > dl_debug on
>> >> >> >
>> >> >> > to get a trace that would tell us anything about that?
>> >> >> >
>> >> >> > Are you using the windows gtalk client or the web site?
>> >> >> > We are following google who have been pushing for the online
>> >> >> > client
>> >> >> > embedded
>> >> >> > in the gmail website.
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> here is the client:
>> >> >> >>
>> >> >> >> <profile type="client">
>> >> >> >> ?<param name="name" value="gmail.com"/>
>> >> >> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> >> >> ?<param name="password" value="password"/>
>> >> >> >> ?<param name="server" value="talk.google.com"/>
>> >> >> >> ?<param name="dialplan" value="XML"/>
>> >> >> >> ?<param name="contex" value="public"/>
>> >> >> >> ?<param name="message" value="FreeSWITCH"/>
>> >> >> >> ?<param name="rtp-ip" value="auto"/>
>> >> >> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> >> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> >> >> ?<param name="auto-login" value="true"/>
>> >> >> >> ?<param name="auto-reply" value="Welcome"/>
>> >> >> >> ?<param name="sasl" value="plain"/>
>> >> >> >> ?<param name="tls" value="true"/>
>> >> >> >> ?<param name="exten" value="5000"/>
>> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> >> >> </profile>
>> >> >> >>
>> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> >> >> > ALL,
>> >> >> >> >
>> >> >> >> > I just upgrade FS to 16992.
>> >> >> >> >
>> >> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
>> >> >> >> > sounds
>> >> >> >> > ok.
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
>> >> >> >> > microseconds
>> >> >> >> > detected!
>> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> >> >> > experience audio problems.
>> >> >> >> >
>> >> >> >> > uname -a
>> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
>> >> >> >> > 08:27:05
>> >> >> >> > UTC
>> >> >> >> > 2008 i686 GNU/Linux
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > I used a script to check the interupts
>> >> >> >> >
>> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> >> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> >> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> >> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> >> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> >> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> >> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> >> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
>> >> >> >> > answer.
>> >> >> >> > I
>> >> >> >> > originate a call from FS, the console blocks, gtalk client ring
>> >> >> >> > and
>> >> >> >> > it
>> >> >> >> > looks ok when I hit answer, but no sound, FS console still
>> >> >> >> > blocks.
>> >> >> >> > a
>> >> >> >> > few seconds later gtalk client shows:
>> >> >> >> >
>> >> >> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> >> >> > network
>> >> >> >> > problem at 5:50 PM. Please try again.
>> >> >> >> > ?Click here to report this to Google.
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > here is a log with bgapi:
>> >> >> >> > http://pastebin.freeswitch.org/12434
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > rev 14696 works on the same server same conf. another server
>> >> >> >> > running
>> >> >> >> > 16958M works at the same time. I have no clue to check, is it
>> >> >> >> > related
>> >> >> >> > to the timer as in 1)? can some help me to take a look? let me
>> >> >> >> > know
>> >> >> >> > if
>> >> >> >> > you need "dl_debug on" logs.
>> >> >> >> >
>> >> >> >> > Thanks.
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > --
>> >> >> > Anthony Minessale II
>> >> >> >
>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >> >> > ClueCon http://www.cluecon.com/
>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >> >
>> >> >> > AIM: anthm
>> >> >> > MSN:anthony_minessale at hotmail.com
>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> > IRC: irc.freenode.net #freeswitch
>> >> >> >
>> >> >> > FreeSWITCH Developer Conference
>> >> >> > sip:888 at conference.freeswitch.org
>> >> >> > iax:guest at conference.freeswitch.org/888
>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> >> > pstn:+19193869900
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From infos at madovsky.org  Wed Mar 17 08:05:21 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 11:05:21 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705><ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
Message-ID: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>

So why not GPL or Gnu Affero public licence ?

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 17, 2010 9:23 AM
Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


> This is where it gets hairy... Yes the MPL says you have to share your 
> modifications with the original author but as far as I can tell you don't 
> have to distribute to cause this clause to go into effect.
>
> Its like this:
>
> You change switch_ivr_originate.c, you HAVE to share those changes with 
> Anthony.
> You create your own mod_kewlshit.c, its yours to do with what you want and 
> you don't have to share.
>
> You change any code that is included with the download YOU MUST share with 
> the original author.
>
> /b
>
>
> On Mar 16, 2010, at 11:30 PM, jay binks wrote:
>
>> Im not on the FS dev team, but I think you may be a little mistaken.
>>
>> generally your free to make modifications to open source code ( for
>> internal use )
>> however its when you distribute the resultant product, then you have
>> to make the source available..
>>
>> if your only using it in-house, I dont believe your required to
>> contribute the code back.
>> ( however your going to end up having issues maintaining your patch
>> set over a long period.. like can be seen here )
>>
>> see here :
>>
>>> must be made available in Source Code form under the terms of this 
>>> License either on the
>>> same media as an Executable version
>>
>> if your not distributing your executable to any 3rd parties... then I
>> dont believe you have to distribute the modified source..
>>
>> Correct me if Im wrong though guys...
>>
>> Jay
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From rupa at rupa.com  Wed Mar 17 08:28:41 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 17 Mar 2010 10:28:41 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
	<b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
	<b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>
Message-ID: <b39273531003170828n32e2e2d6ic7170afcc09e6c1e@mail.gmail.com>

Thanks

On Wed, Mar 17, 2010 at 9:33 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Jira opened. It's for my first time so may be not perfect.
>
> --
-Rupa
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From anthony.minessale at gmail.com  Wed Mar 17 08:31:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 10:31:44 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
	<191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
	<23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>
Message-ID: <191c3a031003170831l2a89137aw98df9d85c109c31@mail.gmail.com>

And trunk also works for us on 2 different tests so I am going to guess its
a configuration issue or some
other variable besides the version of the software.

mod_dingaling has not really changed in months.

We cannot reproduce it so if you want me to fix it you would have to let me
into your box when I have time.
our key is in the source tree in support-d/shinzon.pub

email me the login details at consulting at freeswitch.org and I'll have a
look.


On Wed, Mar 17, 2010 at 10:03 AM, Seven Du <dujinfang at gmail.com> wrote:

> No.
>
> FreeSWITCH runs on public IP.  my gtalk client behind NAT.  As I said,
> old versions(e.g. 14696) work on the same server with the same config.
>
> 2010/3/17 Anthony Minessale <anthony.minessale at gmail.com>:
> > Were you using gtalk and FS both behind nat with the public gtalk server
> by
> > any chance?
> > This setup requires special configuration.
> >
> >
> > On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > Are you bridging to sip or some ivr and making dingaling wait until
> the
> >> > call
> >> > answers?
> >> >
> >>
> >> no I was originate dingaling/... &echo,   also tried originate
> >> dingaling/...  extension XML public (which will hit dialplan and
> >> answer() after client answers), neither work.
> >>
> >>
> >> > gtalk is sort of missing key telephony features like the concept of
> pdd
> >> > and
> >> > indication progress etc.
> >> >
> >> > Try answering the call in your dialplan before you bridge.
> >> >
> >> > I think the gtalk client is getting impatient because you did not
> answer
> >> > soon enough with an ip choice
> >> > so it starts sending all these extra ones at you and gave up on the
> one
> >> > we
> >> > chose.
> >> >
> >> >
> >>
> >> I tested on another server runing FreeSWITCH Version 1.0.trunk
> >> (16958M), it works.
> >>
> >> see http://pastebin.freeswitch.org/12463
> >>
> >> interesting, messages like the same, but I see the working version has
> >> one more line on line 174:
> >>
> >> http://pastebin.freeswitch.org/12463 line: 173 to 175
> >> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
> >> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
> >> [dingaling/gmail.com/seven at idapted.com] has been answered
> >> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
> >> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> >> CS_ROUTING
> >>
> >>
> >> http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)
> >>
> >> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
> >> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
> >> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> >> CS_ROUTING
> >>
> >> Do you see any other difference? Thanks.
> >>
> >>
> >> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> > FS would prefer you run your kernel at 1000hz you are probably on a
> >> >> > VM
> >> >> > if
> >> >> > its 10000hz
> >> >> >
> >> >>
> >> >> It's a  physical machine. And I'm also running 20 skype clients. as
> >> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> >> >> 250 HZ. weird.
> >> >>
> >> >>
> >> >> here is the dingaling log with dl_debug on
> >> >>
> >> >> http://pastebin.freeswitch.org/12453
> >> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
> >> >>
> >> >>
> >> >> still get this message on client:
> >> >>
> >> >> Sorry! The voice chat with language_lesson failed because of a
> network
> >> >> problem at 5:39 PM. Please try again.
> >> >>  Click here to report this to Google
> >> >>
> >> >>
> >> >>
> >> >> Thanks.
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> >
> >> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> >> >> Windows client and web site on Mac. They both work on the other
> two
> >> >> >> instances of FreeSWITCH I mentioned in this thread.
> >> >> >>
> >> >> >> btw, can you give me a hint on the timer warning?
> >> >> >>
> >> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> >> > you would have to execute
> >> >> >> >
> >> >> >> > dl_debug on
> >> >> >> >
> >> >> >> > to get a trace that would tell us anything about that?
> >> >> >> >
> >> >> >> > Are you using the windows gtalk client or the web site?
> >> >> >> > We are following google who have been pushing for the online
> >> >> >> > client
> >> >> >> > embedded
> >> >> >> > in the gmail website.
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> here is the client:
> >> >> >> >>
> >> >> >> >> <profile type="client">
> >> >> >> >>  <param name="name" value="gmail.com"/>
> >> >> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >> >> >>  <param name="password" value="password"/>
> >> >> >> >>  <param name="server" value="talk.google.com"/>
> >> >> >> >>  <param name="dialplan" value="XML"/>
> >> >> >> >>  <param name="contex" value="public"/>
> >> >> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >> >> >>  <param name="rtp-ip" value="auto"/>
> >> >> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >> >> >>  <param name="auto-login" value="true"/>
> >> >> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >> >> >>  <param name="sasl" value="plain"/>
> >> >> >> >>  <param name="tls" value="true"/>
> >> >> >> >>  <param name="exten" value="5000"/>
> >> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/>
> -->
> >> >> >> >> </profile>
> >> >> >> >>
> >> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> >> >> > ALL,
> >> >> >> >> >
> >> >> >> >> > I just upgrade FS to 16992.
> >> >> >> >> >
> >> >> >> >> > 1) timer warning: How's possible the 10000 microseconds?
> audio
> >> >> >> >> > sounds
> >> >> >> >> > ok.
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
> >> >> >> >> > microseconds
> >> >> >> >> > detected!
> >> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You
> may
> >> >> >> >> > experience audio problems.
> >> >> >> >> >
> >> >> >> >> > uname -a
> >> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
> >> >> >> >> > 08:27:05
> >> >> >> >> > UTC
> >> >> >> >> > 2008 i686 GNU/Linux
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > I used a script to check the interupts
> >> >> >> >> >
> >> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> >> >> >> >> > answer.
> >> >> >> >> > I
> >> >> >> >> > originate a call from FS, the console blocks, gtalk client
> ring
> >> >> >> >> > and
> >> >> >> >> > it
> >> >> >> >> > looks ok when I hit answer, but no sound, FS console still
> >> >> >> >> > blocks.
> >> >> >> >> > a
> >> >> >> >> > few seconds later gtalk client shows:
> >> >> >> >> >
> >> >> >> >> > Sorry! The voice chat with language_lesson failed because of
> a
> >> >> >> >> > network
> >> >> >> >> > problem at 5:50 PM. Please try again.
> >> >> >> >> >  Click here to report this to Google.
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > here is a log with bgapi:
> >> >> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > rev 14696 works on the same server same conf. another server
> >> >> >> >> > running
> >> >> >> >> > 16958M works at the same time. I have no clue to check, is it
> >> >> >> >> > related
> >> >> >> >> > to the timer as in 1)? can some help me to take a look? let
> me
> >> >> >> >> > know
> >> >> >> >> > if
> >> >> >> >> > you need "dl_debug on" logs.
> >> >> >> >> >
> >> >> >> >> > Thanks.
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > --
> >> >> >> > Anthony Minessale II
> >> >> >> >
> >> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> >> > ClueCon http://www.cluecon.com/
> >> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >> >
> >> >> >> > AIM: anthm
> >> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >> >
> >> >> >> > FreeSWITCH Developer Conference
> >> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >> > pstn:+19193869900
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Tue Mar 16 15:21:45 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Tue, 16 Mar 2010 22:21:45 +0000
Subject: [Freeswitch-users] How long did the phone ring?
Message-ID: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>

I'm converting a call-center app from Asterisk to FreeSwitch (using xml and
javascript dialplans) and I think I've worked out how to do nearly
everything, except for tracking one important metric: How long the phone
rang before an agent picked it up.

I'm not even entirely sure what to call it in FreeSwitch lingo - so that
would be a good starting point :-)  (Media-duration maybe?)

I can work out the length of ringing for calls that didn't connect, by
logging when the bridge is attempted, and after it fails and comparing the
time. But if the person being called picks up, then all I know from that is
the total length of the call including the ringing/media time, not the media
time and connected time as 2 separate values

The info application gives several useful looking variables like:
Caller-Channel-Progress-Media-Time
Caller-Channel-Transfer-Time
...but they're still set to zero.

I thought I might be able to subtract  Caller-Channel-Created-Time  from
Caller-Channel-Answered-Time  to get the milliseconds of how long the phone
rang for - but there's no reliable correlation between the difference and
how long the phone rang.

In asterisk I achieved this by logging when I was about to make a call, then
running a macro when the Dial() connected, like so:
    exten => s,n,Macro(Dial_Attempted)
    exten => s,n,Dial(SIP/1001,20,M(Dial_Succeeded))
    exten => s,n,Macro(Dial_Failed)  ;Only gets here if the Dial didn't
connect.

Is there any way to do this in FreeSwitch?

Cheers,
Fraser
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From cucku.cucku at yahoo.com.vn  Wed Mar 17 08:29:17 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Wed, 17 Mar 2010 23:29:17 +0800 (SGT)
Subject: [Freeswitch-users] Newbie question on freeswitch
Message-ID: <130309.20630.qm@web76211.mail.sg1.yahoo.com>

HI list
i am new to Freeswitch, i am successfully install freeswitchand i use the default configi use 1000 and 1001 to register successfull and make call between 2 endpoints fine
after that, i create a new 8000.xml file which in the same location as 1000.xmlfreeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml<include>??<user id="8000">?? ?<params>?? ? ?<param name="password" value="$${default_password}"/>?? ? ?<param name="vm-password" value="1000"/>?? ?</params>?? ?<variables>?? ? ?<variable name="toll_allow" value="domestic,international,local"/>?? ? ?<variable name="accountcode" value="8000"/>?? ? ?<variable name="user_context" value="default"/>?? ? ?<variable name="effective_caller_id_name" value="Extension 8000"/>?? ? ?<variable name="effective_caller_id_number" value="8000"/>?? ? ?<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>?? ? ?<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>?? ? ?<variable name="callgroup" value="techsupport"/>?? ?</variables>??</user></include>
the 8000 is registered successfull but i cannot use the 1000 call to 8000
please helpHa`


      C?m gi?c an to?n h?n khi online. N?ng c?p l?n Internet Explorer 8 m?i, an to?n h?n,t?i ?u h?a cho Yahoo!. Ho?n to?n mi?n ph?. T?i IE8 t?i ??y! 
http://downloads.yahoo.com/vn/internetexplorer/
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From lloyd.aloysius at gmail.com  Wed Mar 17 08:43:55 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 11:43:55 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
	<82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>
Message-ID: <8a19bf2e1003170843ub5bbec5pcc6773d2bbe8f646@mail.gmail.com>

Rupa & Tom , Thank you for the suggestion.

Lloyd

On Tue, Mar 16, 2010 at 10:18 PM, Tom Christensen <pavera at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_fifo#Dialplan_Example
> At least with mod_fifo (the call parking solution I'm using) there is a
> pretty clear example in the wiki.
>
> On Tue, Mar 16, 2010 at 6:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> when a user parked to a slot how to handle the park time out. Is there any
>> way to transfer to the call back to a receptionist.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>>
>>>> Hello,
>>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>>> is really nice, straightforward, and I appreciate the design of the system.
>>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>>> really looking for something more stable, and I think I've found it here!
>>>>
>>>> One issue I've run into, and I'm not sure if its just that my use case
>>>> is abnormal, hence the email.
>>>>
>>>> I've worked with PBX systems for a number of years (11 now...), and in
>>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>>> feature.  By this I mean, you think party X (the intended recipient of the
>>>> call) is in the building, but they are not at their desk.  So, you park the
>>>> call, and then page, or by some other means attempt to locate party X and
>>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>>> party X or you can't find them in time, the call should time out back to the
>>>> parking party (receptionist) to be handled appropriately (message taken,
>>>> sent to voicemail, transferred to party X's boss)
>>>>
>>>> This scenario requires a few things:
>>>> 1) the receptionist/person who actually receives the call needs to be
>>>> able to put the call in a specific place (extension) that doesn't change
>>>> 2) Party X (the intended recipient) needs to be able to access that
>>>> specific place (extension) from any phone inside the company
>>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>>> party X and one for party Y, in that order, but party Y gets to a phone
>>>> first, they will be connected to party X's caller)
>>>> 4) The receptionist has to be protected from or provided with sufficient
>>>> information to avoid connecting 2 customers with each other (IE the
>>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>>> there...
>>>>
>>>> The closest I can get to implementing the above scenario is by using
>>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>>> receptionist phone to "notify" her that someone is already parked in a
>>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>>> system, and if she fat fingers her transfer, she will double up a queue,
>>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>>> becomes impossible to appropriately handle call park.  If the receptionist
>>>> is away from her desk and attempts to answer a call, she can't tell which
>>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>>> users are prohibited from using call park as they cannot know which slots
>>>> are free (the need for this arises often in certain industries, such as law,
>>>> client is speaking with attorney, attorney finishes and says "let me connect
>>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>>> you.", paralegal is not at desk, atty needs to park call and page his
>>>> assistant, maybe they are in the copy room, or the library doing
>>>> research...)
>>>>
>>>> The asterisk parking lot seems to solve this problem quite elegantly
>>>> (although the stability issues of asterisk undermine it).  It is simple to
>>>> use (receptionist and everyone else only needs to remember 1 extension to
>>>> park any call), The callers/parking spots are maintained by the system to
>>>> prevent double parking, and each caller sits at a specific extension until
>>>> they are picked up or time out.
>>>>
>>>> So, the question boils down to how do people using freeswitch view/use
>>>> call park?  Is there a "better" roaming feature that I just don't know
>>>> about?  If it is just something that hasn't been implemented/thought about
>>>> this way yet, any idea what it would take to get a call parking feature that
>>>> satisfies these requirements into freeswitch?
>>>>
>>>> Thanks for your time, sorry for the long email, hopefully it explains
>>>> well enough what I'm trying to accomplish.
>>>>
>>>> -Tom
>>>>
>>>
>>> Tom,
>>>
>>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>>> Secondly, thank you for actually looking at the documentation before asking
>>> your question.
>>>
>>> You've got yourself quite the little conundrum. Many of us use park in a
>>> more limited role, or we rely on the receptionist not to do silly things
>>> like sending one external caller to the slot of another external caller. We
>>> all agree that's a bad idea, unless it is done deliberately to connect two
>>> external parties. It sounds to me like this particular situation needs an
>>> operator panel app where the receptionist can have a pretty web page showing
>>> what calls are parked where. To my knowledge, this does not yet exist in
>>> FreeSWITCH. The good news is that FS provides all the requisite building
>>> blocks to create such an application using the event socket. How do you feel
>>> about doing a little development? :)
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Mar 17 08:46:49 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 10:46:49 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
Message-ID: <85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>

Call start answer time minus call start time = ring time in the CDR

/b

On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:

> I'm converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I've worked out how to do nearly everything, except for tracking one important metric: How long the phone rang before an agent picked it up.



From jmesquita at freeswitch.org  Wed Mar 17 08:50:25 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 12:50:25 -0300
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
Message-ID: <ebbc2d551003170850v2a0ab9fagb099cb9425fe78b3@mail.gmail.com>

Look at the Local_Extension in dialplan/default.xml

The regex doesn't match 8000 so you can call from one to another. All you
gotta do is change that and issue a reloadxml on fs_cli.

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 12:29 PM, false <cucku.cucku at yahoo.com.vn> wrote:

> HI list
>
> i am new to Freeswitch, i am successfully install freeswitch
> and i use the default config
> i use 1000 and 1001 to register successfull and make call between 2
> endpoints fine
>
> after that, i create a new 8000.xml file which in the same location as
> 1000.xml
> freeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml
> <include>
>   <user id="8000">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="vm-password" value="1000"/>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="8000"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 8000"/>
>       <variable name="effective_caller_id_number" value="8000"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
> the 8000 is registered successfull but i cannot use the 1000 call to 8000
>
> please help
> Ha`
>
> B?n c? blog tr?n Yahoo! 360plus ch?a?
> ??ng k?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Wed Mar 17 08:51:43 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 11:51:43 -0400
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
Message-ID: <8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>

Adjust the dial plan to call extension  8000. The default Dial plan like
below

<condition field="destination_number" expression="^(10[01][0-9])$">

change to

<condition field="destination_number" expression="^(*[1-8]*0[01][0-9])$">

then reloadxml

Hope this help you.

Lloyd

On Wed, Mar 17, 2010 at 11:29 AM, false <cucku.cucku at yahoo.com.vn> wrote:

> HI list
>
> i am new to Freeswitch, i am successfully install freeswitch
> and i use the default config
> i use 1000 and 1001 to register successfull and make call between 2
> endpoints fine
>
> after that, i create a new 8000.xml file which in the same location as
> 1000.xml
> freeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml
> <include>
>   <user id="8000">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="vm-password" value="1000"/>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="8000"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 8000"/>
>       <variable name="effective_caller_id_number" value="8000"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
> the 8000 is registered successfull but i cannot use the 1000 call to 8000
>
> please help
> Ha`
>
> B?n c? blog tr?n Yahoo! 360plus ch?a?
> ??ng k?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at fowler.cc  Wed Mar 17 08:53:09 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 17 Mar 2010 11:53:09 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>

I was thrilled when mod_silk was released... but my joy was short lived when I got this reply from Skype about SILK support on their "Skype for SIP beta".


"We have double checked this with Skype for SIP developers.  They said that at the moment SILK codecs are not yet supported, even though we have information about this on the web site.



Most probably SILK codecs will be supported when Skype for SIP goes out from beta stage."



Just wanted to share to avoid frustration for other FreeSWITCHers.

Chris.

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From brian at freeswitch.org  Wed Mar 17 08:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 10:56:56 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
Message-ID: <8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>

Thats funny their own internal team doesn't implement the codec as quickly as we did?  har har har

/b

On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:

> Just wanted to share to avoid frustration for other FreeSWITCHers.

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From brian at freeswitch.org  Wed Mar 17 09:20:10 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 11:20:10 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
Message-ID: <18E6B133-7DF4-4079-9D33-1004FF501379@freeswitch.org>

I just updated our code and module to the official release source from Skype.  They did make a few modifications.

/b

On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:

> I was thrilled when mod_silk was released? but my joy was short lived when I got this reply from Skype about SILK support on their ?Skype for SIP beta?.
>  
> ?We have double checked this with Skype for SIP developers.  They said that at the moment SILK codecs are not yet supported, even though we have information about this on the web site.
>  
> Most probably SILK codecs will be supported when Skype for SIP goes out from beta stage.?
>  
>  
> Just wanted to share to avoid frustration for other FreeSWITCHers.
>  
> Chris.

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From infos at madovsky.org  Wed Mar 17 09:29:56 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 12:29:56 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
	<8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>
Message-ID: <3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>

I think when you have millions of users they need more attention before to release anything.
I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
so it's better to be sure that the bug is well corrected...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 11:56 AM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Thats funny their own internal team doesn't implement the codec as quickly as we did?  har har har


  /b


  On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:


    Just wanted to share to avoid frustration for other FreeSWITCHers.




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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Mar 17 09:33:59 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 11:33:59 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
	<8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>
	<3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
Message-ID: <E367901D-1D54-4BB8-80A3-3EB86D79333B@freeswitch.org>

Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P

/b

On Mar 17, 2010, at 11:29 AM, Madovsky wrote:

> I think when you have millions of users they need more attention before to release anything.
> I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
> so it's better to be sure that the bug is well corrected...

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From infos at madovsky.org  Wed Mar 17 09:44:11 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 12:44:11 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net><8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org><3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
	<E367901D-1D54-4BB8-80A3-3EB86D79333B@freeswitch.org>
Message-ID: <856BB38F378B4E5DBD047D4AC90D9E62@MOBILEE1705>

I wish also for you guys...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 12:33 PM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P


  /b


  On Mar 17, 2010, at 11:29 AM, Madovsky wrote:


    I think when you have millions of users they need more attention before to release anything.
    I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
    so it's better to be sure that the bug is well corrected...




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Mar 17 09:57:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 11:57:00 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
Message-ID: <191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>

even better you have the progress and progress_media timestamps too
so you can measure from the instance you got the first ringing indication
so then you can also measure how long it took to start ringing too (PDD)


On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org> wrote:

> Call start answer time minus call start time = ring time in the CDR
>
> /b
>
> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>
> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
> and javascript dialplans) and I think I've worked out how to do nearly
> everything, except for tracking one important metric: How long the phone
> rang before an agent picked it up.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 10:01:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 12:01:46 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
	<2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
Message-ID: <191c3a031003171001l3b2f5911v6e47402419781819@mail.gmail.com>

We don't really care what he does with the code.
But if he expects us to help him with a forked tree, it's not going to
happen.

We expect people who make any meaningful changes to the FS core to be wise
enough to work with
us to keep the code in tree so the community can continue to maintain it and
so it can grow with the rest of the
project.

The GPL license is a whole different topic I will avoid at the moment.


On Wed, Mar 17, 2010 at 10:05 AM, Madovsky <infos at madovsky.org> wrote:

> So why not GPL or Gnu Affero public licence ?
>
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, March 17, 2010 9:23 AM
> Subject: Re: [Freeswitch-users] freeswitch to cisco gateway
>
>
> > This is where it gets hairy... Yes the MPL says you have to share your
> > modifications with the original author but as far as I can tell you don't
> > have to distribute to cause this clause to go into effect.
> >
> > Its like this:
> >
> > You change switch_ivr_originate.c, you HAVE to share those changes with
> > Anthony.
> > You create your own mod_kewlshit.c, its yours to do with what you want
> and
> > you don't have to share.
> >
> > You change any code that is included with the download YOU MUST share
> with
> > the original author.
> >
> > /b
> >
> >
> > On Mar 16, 2010, at 11:30 PM, jay binks wrote:
> >
> >> Im not on the FS dev team, but I think you may be a little mistaken.
> >>
> >> generally your free to make modifications to open source code ( for
> >> internal use )
> >> however its when you distribute the resultant product, then you have
> >> to make the source available..
> >>
> >> if your only using it in-house, I dont believe your required to
> >> contribute the code back.
> >> ( however your going to end up having issues maintaining your patch
> >> set over a long period.. like can be seen here )
> >>
> >> see here :
> >>
> >>> must be made available in Source Code form under the terms of this
> >>> License either on the
> >>> same media as an Executable version
> >>
> >> if your not distributing your executable to any 3rd parties... then I
> >> dont believe you have to distribute the modified source..
> >>
> >> Correct me if Im wrong though guys...
> >>
> >> Jay
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Wed Mar 17 10:11:36 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 13:11:36 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
Message-ID: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>

But Brian,

are you ready to answer to 40 millions users if  the bug is not resolved ? :D
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 12:44 PM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  I wish also for you guys...
    ----- Original Message ----- 
    From: Brian West 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 17, 2010 12:33 PM
    Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


    Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P 


    /b


    On Mar 17, 2010, at 11:29 AM, Madovsky wrote:


      I think when you have millions of users they need more attention before to release anything.
      I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
      so it's better to be sure that the bug is well corrected...




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


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Mar 17 10:15:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 12:15:52 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
Message-ID: <6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>

Sure... I would love that.

/b

On Mar 17, 2010, at 12:11 PM, Madovsky wrote:

> But Brian,
>  
> are you ready to answer to 40 millions users if  the bug is not resolved ? :D
> 

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From maciej.aniserowicz at gmail.com  Wed Mar 17 10:17:46 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 17 Mar 2010 09:17:46 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
Message-ID: <1268846266197-4751694.post@n2.nabble.com>


Hi Brian,
No, it's not trunk, I updated ~2 weeks ago.

Is this a known issue and updating to trunk will fix this problem?

I tried Anthony's suggestion with waiting for park AND answer events to
begin call processing... and I pasted another scenario. Call are bridged
correcly and can talk to each other, but hanging up 2nd call ends up in
hanging up the first one with ORIGINATOR_CANCEL cause...
Here it is, < 290 lines of commented logs:
http://pastebin.freeswitch.org/12468


-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751694.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fraserredmond at gmail.com  Wed Mar 17 10:17:54 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 17:17:54 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
Message-ID: <c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>

Yeah, that's what I had expected, and was why I was confused.

I've done some more tests, and tracked the results a little closer in a
spreadsheet, and it seems that when I call the extension directly from
another sipphone the  Caller-Channel-Answered-Time minus
Caller-Channel-Created-Time  matches the ring time, so that scenario is
fine.

When I call into an IVR with the javascript dialplan and then create the new
session and bridge them that way, the Answered-Time and Created-Time, that
are reported after the call ends are reported on the A-leg's Created &
Answered.

The good news is that the Progress-Time is reported on when the B-leg
started ringing, so I can know when the call started ringing.
The bad news is that the Progress-Media-Time is always blank

I can take the Progress-Time and compare it to the system-clock, and that
should generally be accurate to a second or so, which is better than
nothing, but it'd be nice to do it properly.

Any other ideas? Is there any channel-variables I can set that would be
worth playing with? I'm currently only setting:
ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true

Cheers,
Fraser




On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> even better you have the progress and progress_media timestamps too
> so you can measure from the instance you got the first ringing indication
> so then you can also measure how long it took to start ringing too (PDD)
>
>
> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Call start answer time minus call start time = ring time in the CDR
>>
>> /b
>>
>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>
>> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
>> and javascript dialplans) and I think I've worked out how to do nearly
>> everything, except for tracking one important metric: How long the phone
>> rang before an agent picked it up.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Mar 17 10:38:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 12:38:34 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268846266197-4751694.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
Message-ID: <3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>

Why don't you update and see... this problem sounds familiar but without you updating I can't 100% recall.

When we say update... be like Nike and JUST DO IT!  ;)

/b

On Mar 17, 2010, at 12:17 PM, Maciej Aniserowicz wrote:

> 
> Hi Brian,
> No, it's not trunk, I updated ~2 weeks ago.
> 
> Is this a known issue and updating to trunk will fix this problem?
> 
> I tried Anthony's suggestion with waiting for park AND answer events to
> begin call processing... and I pasted another scenario. Call are bridged
> correcly and can talk to each other, but hanging up 2nd call ends up in
> hanging up the first one with ORIGINATOR_CANCEL cause...
> Here it is, < 290 lines of commented logs:
> http://pastebin.freeswitch.org/12468

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From msc at freeswitch.org  Wed Mar 17 10:50:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 10:50:49 -0700
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
References: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
Message-ID: <87f2f3b91003171050h5d791ab7t409d39c43d1abdf2@mail.gmail.com>

On Sun, Mar 14, 2010 at 1:01 PM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> > Pass them as arguments to your luarun hangup hook.
> And what about variables are not available before the call setup (like
> uuid), only during the call progress or after the hangup???
> I would need to access the originate_disposition variable...
>
> >> 'luarun hangup.lua ${originate_disposition}'
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> any other hint???
>

Use the magic env object, which is an event object with all the chan vars
for the channel that just hung up. More info here:
http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object
-MC
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From maciej.aniserowicz at gmail.com  Wed Mar 17 10:59:00 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 17 Mar 2010 09:59:00 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
Message-ID: <1268848740857-4751927.post@n2.nabble.com>


OK, call me Nike;). I'll do that in the upcoming days and see if the problem
stays.

The reason I'm not downloading trunk very frequently is that almost every
time I update, something just stops working. So once I am in a
"pretty-stable" state I don't think about FS update for some time.
And, well, this last update wasn't different from the previous ones as you
can see :).

I'll post a message in this subject to let you know if it works with the
latest bits.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751927.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Wed Mar 17 11:11:08 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 13:11:08 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268848740857-4751927.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
Message-ID: <90C5E5AA-C6A5-48BF-9C73-C1807676D9DE@freeswitch.org>


On Mar 17, 2010, at 12:59 PM, Maciej Aniserowicz wrote:

> 
> OK, call me Nike;). I'll do that in the upcoming days and see if the problem
> stays.
> 
> The reason I'm not downloading trunk very frequently is that almost every
> time I update, something just stops working. So once I am in a
> "pretty-stable" state I don't think about FS update for some time.

I suspect you're doing a dirty update and have build skew.  How exactly are you doing your updates?  Trunk has been very stable and only getting better moving forward... if you had real issues you should be reporting them in jira.  Not ignoring the issues.  Thats how things don't get fixed.  I would rather you open a jira and let me see the issue and close it with "Not A Bug" then chance letting a bug slip thru.

> And, well, this last update wasn't different from the previous ones as you
> can see :).
> 
> I'll post a message in this subject to let you know if it works with the
> latest bits.
> -- 



From anthony.minessale at gmail.com  Wed Mar 17 11:31:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:31:39 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268848740857-4751927.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
Message-ID: <191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>

What If I used that same policy that you use to update when it came to
answering your emails?

Please update to latest trunk so I can confirm your traces against the
latest code.
Also run the command show channels while it's bridged.

I would prefer if you do it ASAP, now that you have us worrying about your
issue, its your duty to see it through.




On Wed, Mar 17, 2010 at 12:59 PM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> OK, call me Nike;). I'll do that in the upcoming days and see if the
> problem
> stays.
>
> The reason I'm not downloading trunk very frequently is that almost every
> time I update, something just stops working. So once I am in a
> "pretty-stable" state I don't think about FS update for some time.
> And, well, this last update wasn't different from the previous ones as you
> can see :).
>
> I'll post a message in this subject to let you know if it works with the
> latest bits.
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751927.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 11:47:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:47:35 -0500
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
Message-ID: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>

side note:

Some of mod_valet_parking seems undocumented you can do the following

 <extension name="park-in">
      <condition field="destination_number" expression="^(8000)$">
        <action application="valet_park" data="auto in 8001 8999"/>
      </condition>
    </extension>


 <extension name="park-out">
     <condition field="destination_number" expression="^(8\d{3})$">
        <action application="answer"/>
        <action application="valet_park" data="test $1"/>
     </condition>
</extension>


So if you attended transfer to 8000, you will park them at the next
available slot
This is missing the blf like fifo has but it will be unique announced slots.


also

you can add this before any call

 <action application="bind_meta_app" data="1 a s valet_park::test auto in
8001 8999"/>

so when on the phone you can dial *1 to park the call to the next free slot
8001-8999




On Tue, Mar 16, 2010 at 9:17 PM, Tom Christensen <pavera at gmail.com> wrote:

>
>
> On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Michael,
> Thanks for the reply. I worked on it more with the business in question
> today and we came up with a hybrid solution which I think will cover all the
> bases.
> 1) I implemented a "call park" extension on each phone, these are standard
> mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
> freeswitch config here BTW, like 8 lines in the dialplan handles this for
> all 50 extensions).  This way everyone has a call park extension that can be
> picked up anywhere, the numbering scheme is essentially take your extension
> prepend a 2, that is your call park extension, so if someone pages you, you
> just pick up any phone and dial 2+ext and that will pick the call up.
>  Everyone in the office can use this feature to park a call for anyone,
> because only calls for a specific person will ever be parked in that queue,
> you can put multiple people there without issue.
> 2) Implemented as mentioned a "parking lot" set of mod_fifo queues that
> have SLA/BLA set up on the receptionist phone.  She can use these queues to
> quickly and easily park a bunch of people at once (IE when there are call
> floods) and then send them off to where they need to go as time permits.  If
> she isn't at her phone, she can still answer calls and park them onto
> individual users park extensions.  Still doesn't have the "safety" of not
> double parking someone in the parking lot, but we'll trust her to be able to
> see red lights :)
>
> This whole thing goes live Thursday, so wish me luck :)
> Thanks again!
> Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 11:52:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:52:23 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
Message-ID: <191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>

progress_media_time is only set if you get a 183
if you get a 180 with no media it uses progress_time instead

as long as you have one or the other you have the timestamp of when it
started ringing.





On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Yeah, that's what I had expected, and was why I was confused.
>
> I've done some more tests, and tracked the results a little closer in a
> spreadsheet, and it seems that when I call the extension directly from
> another sipphone the  Caller-Channel-Answered-Time minus
> Caller-Channel-Created-Time  matches the ring time, so that scenario is
> fine.
>
> When I call into an IVR with the javascript dialplan and then create the
> new session and bridge them that way, the Answered-Time and Created-Time,
> that are reported after the call ends are reported on the A-leg's Created &
> Answered.
>
> The good news is that the Progress-Time is reported on when the B-leg
> started ringing, so I can know when the call started ringing.
> The bad news is that the Progress-Media-Time is always blank
>
> I can take the Progress-Time and compare it to the system-clock, and that
> should generally be accurate to a second or so, which is better than
> nothing, but it'd be nice to do it properly.
>
> Any other ideas? Is there any channel-variables I can set that would be
> worth playing with? I'm currently only setting:
> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> even better you have the progress and progress_media timestamps too
>> so you can measure from the instance you got the first ringing indication
>> so then you can also measure how long it took to start ringing too (PDD)
>>
>>
>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Call start answer time minus call start time = ring time in the CDR
>>>
>>> /b
>>>
>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>
>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
>>> and javascript dialplans) and I think I've worked out how to do nearly
>>> everything, except for tracking one important metric: How long the phone
>>> rang before an agent picked it up.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From lloyd.aloysius at gmail.com  Wed Mar 17 11:54:41 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 14:54:41 -0400
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
Message-ID: <8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>

 <action application="bridge" data="user/${dialed_extension}@
${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
mber}"/>

I setup the above dial plan. The call to extension works perfect. But It
never call to the ${followme_neumber}.

How can I fix this issue.

Thanks
Lloyd




On Mon, Mar 8, 2010 at 7:24 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Michael,
>
> Thank you for the suggestion. This suggestion make sense and a proper way
> to implement the follow me. .
>
> Thanks
> Lloyd
>
> On Mon, Mar 8, 2010 at 6:10 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>>
>>> Hi All,
>>>
>>> I would like to setup follow me dialplan for individual extensions. Some
>>> extensions need a follow-me and some not. If the the follow-me not reach any
>>> live person the call should go to the FreeSWTICH voicemail.I have the
>>> extensions range from 201 -209. Also I am using the following Default dial
>>> plan to call between extensions.
>>>
>>> <extension name="Local_Extension">
>>>       <condition field="destination_number" expression="^(20[0-9])$">
>>> <action application="set" data="dialed_extension=$1"/>
>>>  <action application="export" data="dialed_extension=$1"/>
>>> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>>>  <action application="bind_meta_app" data="1 b s execute_extension::dx
>>> XML features"/>
>>> <action application="bind_meta_app" data="2 b s
>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>  <action application="bind_meta_app" data="3 b s execute_extension::cf
>>> XML features"/>
>>> <action application="set" data="ringback=${us-ring}"/>
>>>  <action application="set" data="transfer_ringback=$${hold_music}"/>
>>> <action application="set" data="call_timeout=30"/>
>>>  <!-- <action application="set"
>>> data="sip_exclude_contact=${network_addr}"/> -->
>>> <action application="set" data="hangup_after_bridge=true"/>
>>>  <!--<action application="set"
>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
>>> -->
>>>  <action application="set" data="continue_on_fail=true"/>
>>> <action application="hash"
>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>  <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>> <action application="set"
>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>> var callgroup)}"/>
>>>  <!--<action application="export"
>>> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
>>> var sip_secure_media)}"/>-->
>>>  <action application="hash"
>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>> <action application="bridge" data="user/${dialed_extension}@
>>> ${domain_name},${followme}"/>
>>>  <action application="answer"/>
>>> <action application="sleep" data="1000"/>
>>>  <action application="voicemail" data="default ${domain_name}
>>> ${dialed_extension}"/>
>>>       </condition>
>>>  </extension>
>>>
>>> I use the wiki samples to make a follow me dial plan for extension 201.
>>> Here is the dial plan.
>>>
>>> <extension name="201_followme">
>>>   <condition field="destination_number" expression="201">
>>>    <action application="set" data="hangup_after_bridge=true"/>
>>>    <action application="set" data="continue_on_fail=true"/>
>>>    <action application="set" data="ignore_early_media=true"/>
>>>    <action application="set" data="call_timeout=10"/>
>>>    <action application="bridge" data="sofia/$${domain}/201"/>
>>>    <action application="set" data="call_timeout=13"/>
>>>    <action application="bridge"
>>> data="sofia/gateway/voicepulse/14161235555" />
>>>    <action application="answer"/>
>>>    <action application="sleep" data="1000"/>
>>>    <action application="voicemail" data="default $${domain} 201"/>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> Call to extension 201 , there are two dialplan matching the request.
>>> Either Local_Extension or follwme_201. Which order freeswtich select?
>>>
>>
>> It will match whichever one comes first in the file. You could, for
>> example, put this 201 extension above the Local_Extension. You could also
>> put a custom chan var in the user directory for those who need the
>> follow-me. Then you could have two conditions, both of which need to match:
>>
>> <extension name="Local, follow me">
>>   <condition field="${followme}" expression="^yes$"/>
>>   <condition field="destination_number expression="^(20\d)$">
>>     <!-- followme stuff here -->
>>     ...
>>   </condition>
>> </extension>
>>
>> <extension name="Local No follow me">
>>   <condition field="${followme}" expression="^$"/>
>>   <condition field="destination_number expression="^(20\d)$">
>>     <!-- normal local extension stuff here -->
>>     ...
>>   </condition>
>> </extension>
>>
>> Then in the user setup in the directory add the followme variable only to
>> those who need it. You can also add a variable for the followme phone number
>> if you want. Inside the <variables> node just add whatever you want:
>>
>>   <variable name="followme" value="yes"/>
>>   <variable name="followme_number" value="14161235555"/>
>> Or even
>>   <variable name="followme_dialstring"
>> value="sofia/gateway/voicepulse/14161235555"/>
>>
>> Then you could use ${followme_number} in your bridge like this:
>>
>>   <action application="bridge" data="sofia/gateway/
>> voicepulse/${followme_number}" />
>>
>> Or ${followme_dialstring} like this:
>>   <action application="bridge" data="${followme_dialstring}" />
>>
>> Make sense? :)
>>
>> Give it a try and report back your success/failure/questions/comments/etc.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Wed Mar 17 12:07:05 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 15:07:05 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
	<191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
Message-ID: <8a19bf2e1003171207q2432918g90830c33afd5bdcc@mail.gmail.com>

Thanks Antony.

Lloyd


On Wed, Mar 17, 2010 at 2:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> side note:
>
> Some of mod_valet_parking seems undocumented you can do the following
>
>  <extension name="park-in">
>       <condition field="destination_number" expression="^(8000)$">
>         <action application="valet_park" data="auto in 8001 8999"/>
>       </condition>
>     </extension>
>
>
>  <extension name="park-out">
>      <condition field="destination_number" expression="^(8\d{3})$">
>         <action application="answer"/>
>         <action application="valet_park" data="test $1"/>
>      </condition>
> </extension>
>
>
> So if you attended transfer to 8000, you will park them at the next
> available slot
> This is missing the blf like fifo has but it will be unique announced
> slots.
>
>
> also
>
> you can add this before any call
>
>  <action application="bind_meta_app" data="1 a s valet_park::test auto in
> 8001 8999"/>
>
> so when on the phone you can dial *1 to park the call to the next free slot
> 8001-8999
>
>
>
>
>
> On Tue, Mar 16, 2010 at 9:17 PM, Tom Christensen <pavera at gmail.com> wrote:
>
>>
>>
>> On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>>
>>>> Hello,
>>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>>> is really nice, straightforward, and I appreciate the design of the system.
>>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>>> really looking for something more stable, and I think I've found it here!
>>>>
>>>> One issue I've run into, and I'm not sure if its just that my use case
>>>> is abnormal, hence the email.
>>>>
>>>> I've worked with PBX systems for a number of years (11 now...), and in
>>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>>> feature.  By this I mean, you think party X (the intended recipient of the
>>>> call) is in the building, but they are not at their desk.  So, you park the
>>>> call, and then page, or by some other means attempt to locate party X and
>>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>>> party X or you can't find them in time, the call should time out back to the
>>>> parking party (receptionist) to be handled appropriately (message taken,
>>>> sent to voicemail, transferred to party X's boss)
>>>>
>>>> This scenario requires a few things:
>>>> 1) the receptionist/person who actually receives the call needs to be
>>>> able to put the call in a specific place (extension) that doesn't change
>>>> 2) Party X (the intended recipient) needs to be able to access that
>>>> specific place (extension) from any phone inside the company
>>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>>> party X and one for party Y, in that order, but party Y gets to a phone
>>>> first, they will be connected to party X's caller)
>>>> 4) The receptionist has to be protected from or provided with sufficient
>>>> information to avoid connecting 2 customers with each other (IE the
>>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>>> there...
>>>>
>>>> The closest I can get to implementing the above scenario is by using
>>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>>> receptionist phone to "notify" her that someone is already parked in a
>>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>>> system, and if she fat fingers her transfer, she will double up a queue,
>>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>>> becomes impossible to appropriately handle call park.  If the receptionist
>>>> is away from her desk and attempts to answer a call, she can't tell which
>>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>>> users are prohibited from using call park as they cannot know which slots
>>>> are free (the need for this arises often in certain industries, such as law,
>>>> client is speaking with attorney, attorney finishes and says "let me connect
>>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>>> you.", paralegal is not at desk, atty needs to park call and page his
>>>> assistant, maybe they are in the copy room, or the library doing
>>>> research...)
>>>>
>>>> The asterisk parking lot seems to solve this problem quite elegantly
>>>> (although the stability issues of asterisk undermine it).  It is simple to
>>>> use (receptionist and everyone else only needs to remember 1 extension to
>>>> park any call), The callers/parking spots are maintained by the system to
>>>> prevent double parking, and each caller sits at a specific extension until
>>>> they are picked up or time out.
>>>>
>>>> So, the question boils down to how do people using freeswitch view/use
>>>> call park?  Is there a "better" roaming feature that I just don't know
>>>> about?  If it is just something that hasn't been implemented/thought about
>>>> this way yet, any idea what it would take to get a call parking feature that
>>>> satisfies these requirements into freeswitch?
>>>>
>>>> Thanks for your time, sorry for the long email, hopefully it explains
>>>> well enough what I'm trying to accomplish.
>>>>
>>>> -Tom
>>>>
>>>
>>> Tom,
>>>
>>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>>> Secondly, thank you for actually looking at the documentation before asking
>>> your question.
>>>
>>> You've got yourself quite the little conundrum. Many of us use park in a
>>> more limited role, or we rely on the receptionist not to do silly things
>>> like sending one external caller to the slot of another external caller. We
>>> all agree that's a bad idea, unless it is done deliberately to connect two
>>> external parties. It sounds to me like this particular situation needs an
>>> operator panel app where the receptionist can have a pretty web page showing
>>> what calls are parked where. To my knowledge, this does not yet exist in
>>> FreeSWITCH. The good news is that FS provides all the requisite building
>>> blocks to create such an application using the event socket. How do you feel
>>> about doing a little development? :)
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> Michael,
>> Thanks for the reply. I worked on it more with the business in question
>> today and we came up with a hybrid solution which I think will cover all the
>> bases.
>> 1) I implemented a "call park" extension on each phone, these are standard
>> mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
>> freeswitch config here BTW, like 8 lines in the dialplan handles this for
>> all 50 extensions).  This way everyone has a call park extension that can be
>> picked up anywhere, the numbering scheme is essentially take your extension
>> prepend a 2, that is your call park extension, so if someone pages you, you
>> just pick up any phone and dial 2+ext and that will pick the call up.
>>  Everyone in the office can use this feature to park a call for anyone,
>> because only calls for a specific person will ever be parked in that queue,
>> you can put multiple people there without issue.
>> 2) Implemented as mentioned a "parking lot" set of mod_fifo queues that
>> have SLA/BLA set up on the receptionist phone.  She can use these queues to
>> quickly and easily park a bunch of people at once (IE when there are call
>> floods) and then send them off to where they need to go as time permits.  If
>> she isn't at her phone, she can still answer calls and park them onto
>> individual users park extensions.  Still doesn't have the "safety" of not
>> double parking someone in the parking lot, but we'll trust her to be able to
>> see red lights :)
>>
>> This whole thing goes live Thursday, so wish me luck :)
>> Thanks again!
>> Tom
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 12:11:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:11:10 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
Message-ID: <87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>

On Wed, Mar 17, 2010 at 5:22 AM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Yes
>
>
You can't. The FIFO system isn't transferring the call to the agent's phone.
Rather, it is ringing the agent's phone and when the agent picks up, it then
bridges the caller to the agent.

What exactly are you trying to accomplish? Just curious. By the way you can
possibly resolve this issue by create an "agent access" extension. The idea
is simple: User A picks up and dials an extension; that extension then
connects User A to the longest waiting caller in the FIFO queue.

Use this example to create your "agent access" extension:
http://wiki.freeswitch.org/wiki/Mod_fifo#Take_a_caller_out_of_a_FIFO_queue

The net result is probably the same: a third phone connects to the caller in
queue. If that caller is the only one in the queue then the agent's phone
that is ringing will stop ringing immediately.

-MC
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From msc at freeswitch.org  Wed Mar 17 12:13:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:13:13 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
	<5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
Message-ID: <87f2f3b91003171213k39cd5431m43f28651f869d943@mail.gmail.com>

On Wed, Mar 17, 2010 at 7:00 AM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> How do I do that in Lua?  session:execute("dtmf","start") doesn't work.

session:execute("start_dtmf")
-MC
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From fraserredmond at gmail.com  Wed Mar 17 12:18:00 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 19:18:00 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
Message-ID: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>

Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
way?

I've realised since my last email that even having the progress_time doesn't
help without the progress_media_time or some way to find when the ringing
stopped - getting the server timestamp wouldn't happen until the call ended,
at which point I can't split the ring time from the talk time.

Surely there can't be something that Asterisk handles well that FreeSwitch
doesn't? ;-)

I'm looking into the CDR now - might have to parse the log files to get the
end time from there :-(

Cheers,
Fraser




On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> progress_media_time is only set if you get a 183
> if you get a 180 with no media it uses progress_time instead
>
> as long as you have one or the other you have the timestamp of when it
> started ringing.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Yeah, that's what I had expected, and was why I was confused.
>>
>> I've done some more tests, and tracked the results a little closer in a
>> spreadsheet, and it seems that when I call the extension directly from
>> another sipphone the  Caller-Channel-Answered-Time minus
>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>> fine.
>>
>> When I call into an IVR with the javascript dialplan and then create the
>> new session and bridge them that way, the Answered-Time and Created-Time,
>> that are reported after the call ends are reported on the A-leg's Created &
>> Answered.
>>
>> The good news is that the Progress-Time is reported on when the B-leg
>> started ringing, so I can know when the call started ringing.
>> The bad news is that the Progress-Media-Time is always blank
>>
>> I can take the Progress-Time and compare it to the system-clock, and that
>> should generally be accurate to a second or so, which is better than
>> nothing, but it'd be nice to do it properly.
>>
>> Any other ideas? Is there any channel-variables I can set that would be
>> worth playing with? I'm currently only setting:
>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> even better you have the progress and progress_media timestamps too
>>> so you can measure from the instance you got the first ringing indication
>>> so then you can also measure how long it took to start ringing too (PDD)
>>>
>>>
>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> Call start answer time minus call start time = ring time in the CDR
>>>>
>>>> /b
>>>>
>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>
>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>> everything, except for tracking one important metric: How long the phone
>>>> rang before an agent picked it up.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Mar 17 12:41:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 14:41:36 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>

i'll try again

progress_media and progress are both the same for your purposes
you will only have exactly one of the 2 set.  its the exact time it started
ringing.

for our sake, if we have either of them we will take the one that is not
zero and call it the progress time.

so you have the created time, the progress time , the answered time and the
hangup time


you always have created time and hangup time but not always progress and
answered

so talk time is (hangup_time - (answered_time unless its 0)) or if
answered_time is 0 there is no talk time
ring time is:
         (answered time unless its zero otherwise hangup_time) -
created_time


edit conf/autoload_configs/cdr_csv.conf.xml

uncomment:

<param name="debug" value="true"/>

and restart FS


now when you hangup all your calls you will see the values you have to work
with in the cdr

if you are tying to do billing inside some js script you are using, stop
now, and go learn about cdrs.





On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?
>
> I've realised since my last email that even having the progress_time
> doesn't help without the progress_media_time or some way to find when the
> ringing stopped - getting the server timestamp wouldn't happen until the
> call ended, at which point I can't split the ring time from the talk time.
>
> Surely there can't be something that Asterisk handles well that FreeSwitch
> doesn't? ;-)
>
> I'm looking into the CDR now - might have to parse the log files to get the
> end time from there :-(
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> progress_media_time is only set if you get a 183
>> if you get a 180 with no media it uses progress_time instead
>>
>> as long as you have one or the other you have the timestamp of when it
>> started ringing.
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> Yeah, that's what I had expected, and was why I was confused.
>>>
>>> I've done some more tests, and tracked the results a little closer in a
>>> spreadsheet, and it seems that when I call the extension directly from
>>> another sipphone the  Caller-Channel-Answered-Time minus
>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>> fine.
>>>
>>> When I call into an IVR with the javascript dialplan and then create the
>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>> that are reported after the call ends are reported on the A-leg's Created &
>>> Answered.
>>>
>>> The good news is that the Progress-Time is reported on when the B-leg
>>> started ringing, so I can know when the call started ringing.
>>> The bad news is that the Progress-Media-Time is always blank
>>>
>>> I can take the Progress-Time and compare it to the system-clock, and that
>>> should generally be accurate to a second or so, which is better than
>>> nothing, but it'd be nice to do it properly.
>>>
>>> Any other ideas? Is there any channel-variables I can set that would be
>>> worth playing with? I'm currently only setting:
>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> even better you have the progress and progress_media timestamps too
>>>> so you can measure from the instance you got the first ringing
>>>> indication
>>>> so then you can also measure how long it took to start ringing too (PDD)
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>
>>>>> /b
>>>>>
>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>
>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>> everything, except for tracking one important metric: How long the phone
>>>>> rang before an agent picked it up.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Wed Mar 17 12:46:48 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 17 Mar 2010 20:46:48 +0100
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <4468a6771003171246t76a58d7erc6d4c3fd48a938c1@mail.gmail.com>

SIP 180 = your phone rings (Ringing message)
SIP 183 + sdp = early media or inband ringing (Session Progress message)

You have both vars to watch for.

Ognjen

On Wed, Mar 17, 2010 at 8:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?
>
> I've realised since my last email that even having the progress_time
> doesn't help without the progress_media_time or some way to find when the
> ringing stopped - getting the server timestamp wouldn't happen until the
> call ended, at which point I can't split the ring time from the talk time.
>
> Surely there can't be something that Asterisk handles well that FreeSwitch
> doesn't? ;-)
>
> I'm looking into the CDR now - might have to parse the log files to get the
> end time from there :-(
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> progress_media_time is only set if you get a 183
>> if you get a 180 with no media it uses progress_time instead
>>
>> as long as you have one or the other you have the timestamp of when it
>> started ringing.
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> Yeah, that's what I had expected, and was why I was confused.
>>>
>>> I've done some more tests, and tracked the results a little closer in a
>>> spreadsheet, and it seems that when I call the extension directly from
>>> another sipphone the  Caller-Channel-Answered-Time minus
>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>> fine.
>>>
>>> When I call into an IVR with the javascript dialplan and then create the
>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>> that are reported after the call ends are reported on the A-leg's Created &
>>> Answered.
>>>
>>> The good news is that the Progress-Time is reported on when the B-leg
>>> started ringing, so I can know when the call started ringing.
>>> The bad news is that the Progress-Media-Time is always blank
>>>
>>> I can take the Progress-Time and compare it to the system-clock, and that
>>> should generally be accurate to a second or so, which is better than
>>> nothing, but it'd be nice to do it properly.
>>>
>>> Any other ideas? Is there any channel-variables I can set that would be
>>> worth playing with? I'm currently only setting:
>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> even better you have the progress and progress_media timestamps too
>>>> so you can measure from the instance you got the first ringing
>>>> indication
>>>> so then you can also measure how long it took to start ringing too (PDD)
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>
>>>>> /b
>>>>>
>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>
>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>> everything, except for tracking one important metric: How long the phone
>>>>> rang before an agent picked it up.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Mar 17 12:46:47 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 15:46:47 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
Message-ID: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>

Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

Regards

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 4:35 PM
  Subject: Re: [Freeswitch-users] voicemail options


  ok, I tried 50 times to leave a message,
  it worked only one time.
  It seems that sendmail needs to be free to send the email....

    ----- Original Message ----- 
    From: devel at thom.fr.eu.org 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 3:59 PM
    Subject: Re: [Freeswitch-users] voicemail options


    Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

    I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

     

    Fran?ois

     

     

    De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
    Envoy? : mardi 16 mars 2010 20:30
    ? : freeswitch-users at lists.freeswitch.org
    Objet : Re: [Freeswitch-users] voicemail options

     

    are you running FS as root or a regular user?
    maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



    On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

    I noticed that sendmail makes a segfault when FS is trying to send email

    and sendmail has some emails locked in folder queue....

     

    Regards

     

    Franck


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




    -- 
    Anthony Minessale II

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

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

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



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


    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From kristian.kielhofner at gmail.com  Wed Mar 17 12:50:57 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 17 Mar 2010 15:50:57 -0400
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <4d15ff861003171250g595d0f0bn3d01386b9d603df@mail.gmail.com>

They are referring to SIP 180 and SIP 183.

It's kind of a mess (it is SIP) but both are non-final response codes
to indicate call progress.  Typically speaking, 180 Ringing is
returned without media and tells the receiving device (FreeSWITCH or
your local SIP phone in this case) to activate a local ringback
generator to inform the caller the remote end is progressing.  180
without media is typically used when calling SIP phones.  They know
that they only actually "ring".

183 is typically used with media.  The media contains audio to tell
the caller what is happening.  Typically this audio sounds just like
ringback (depending on region) but it could just as easily be a
ringback tone (Britney Spears or anything else), audio from the remote
end, or just about anything else the caller might need to know (even
an error message).

Most carriers *should* consistently give you 183 with media while the
call progresses, whether the remote end is actually ringing or not
depends.  You have to listen to the audio :).

BTW, as are most things with SIP these are generalities and many not
happen in your environment, most stacks, for example, will handle
180/183 interchangeably - it's the presence of media (SDP) that
matters.

FreeSWITCH has a multitude of options to deal with these scenarios.
First things first - what does your carrier return?

On Wed, Mar 17, 2010 at 3:18 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From robert.dyck at shaw.ca  Wed Mar 17 12:51:55 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 12:51:55 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003170049.06565.robert.dyck@shaw.ca>
	<ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
Message-ID: <201003171251.57460.robert.dyck@shaw.ca>

I can see a process for fscomm. Should I be seeing a process for the core.

When starting fscomm it reports -
Initializing globals... 

Initializing core... 

Everything OK, Entering runtime loop ...

BTW, not Windows but Fedora 11, KDE flavour

Thanks for looking into this
RD


On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> Rob, I believe that the core has failed to load. I have not added any type
> of check on the code level for those cases yet as I believed it would never
> really fail to load. Those cases have been more frequent than what I
> originally expected so I will add that to code today (hopefully later
> tonight).
> 
> Meanwhile, what you can do is check if the ports are open or not on the OS.
> I am not sure how to do that on Windows. Or, maybe we already have logs on
> the log folder?
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > Is there something wrong with my build?
> > 
> > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > [Socket
> > Connection Error]" with fscomm running.
> > 
> > RD
> > 
> > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > With FSComm running you can use fs_cli to try pa list. If u can't list
> > 
> > the
> > 
> > > devices that would be the problem. I'm going to try to dig into this
> > > further on 1 of my desktops w/ pulseaudio.
> > > 
> > > -- W
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From anthony.minessale at gmail.com  Wed Mar 17 12:55:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 14:55:00 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
Message-ID: <191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>

could you get a svn diff of the switch_utils.c so we can  see what the
change was?



On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

>  Finally I succeed to send voicemail with sendmail through a little php
> script (could be Perl also)
> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
> doesn't exist.
>
> I attached my php script and switch_utils.c. Not sure if the FS developers
> will accept my modification
>
> Regards
>
> Franck
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 4:35 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
> ok, I tried 50 times to leave a message,
> it worked only one time.
> It seems that sendmail needs to be free to send the email....
>
>
> ----- Original Message -----
> *From:* devel at thom.fr.eu.org
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 3:59 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
>  Are you running debian, I have the same problem on debian x64 (already
> mentioned here
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
> and cannot fix it.
>
> I don?t know if it?s a priviledge problem, as I got the same results
> running FS with or without ?u.
>
>
>
> Fran?ois
>
>
>
>
>
> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
> Minessale
> *Envoy? :* mardi 16 mars 2010 20:30
> *? :* freeswitch-users at lists.freeswitch.org
> *Objet :* Re: [Freeswitch-users] voicemail options
>
>
>
> are you running FS as root or a regular user?
> maybe there is a permission or resource limitation problem preventing it
> from getting the larger stack space necessary?
>
>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>
> I noticed that sendmail makes a segfault when FS is trying to send email
>
> and sendmail has some emails locked in folder queue....
>
>
>
> Regards
>
>
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Wed Mar 17 12:58:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:58:45 -0700
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
	<191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
Message-ID: <87f2f3b91003171258x77678688v2515af8cc7b62b29@mail.gmail.com>

FYI,

This method and the bind meta app method are now in the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Valet_Autoselect_Parking_Stall

-MC

On Wed, Mar 17, 2010 at 11:47 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> side note:
>
> Some of mod_valet_parking seems undocumented you can do the following
>
>  <extension name="park-in">
>       <condition field="destination_number" expression="^(8000)$">
>         <action application="valet_park" data="auto in 8001 8999"/>
>       </condition>
>     </extension>
>
>
>  <extension name="park-out">
>      <condition field="destination_number" expression="^(8\d{3})$">
>         <action application="answer"/>
>         <action application="valet_park" data="test $1"/>
>      </condition>
> </extension>
>
>
> So if you attended transfer to 8000, you will park them at the next
> available slot
> This is missing the blf like fifo has but it will be unique announced
> slots.
>
>
> also
>
> you can add this before any call
>
>  <action application="bind_meta_app" data="1 a s valet_park::test auto in
> 8001 8999"/>
>
> so when on the phone you can dial *1 to park the call to the next free slot
> 8001-8999
>
>
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From fraserredmond at gmail.com  Wed Mar 17 13:10:33 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 20:10:33 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
Message-ID: <c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>

Thanks, I suspect this is what I really needed:

> if you are tying to do billing inside some js script you are using, stop
now, and go learn about cdrs.

I guess there's more for me to un-learn from Asterisk than I'd thought :-(

Cheers,
Fraser




On Wed, Mar 17, 2010 at 7:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i'll try again
>
> progress_media and progress are both the same for your purposes
> you will only have exactly one of the 2 set.  its the exact time it started
> ringing.
>
> for our sake, if we have either of them we will take the one that is not
> zero and call it the progress time.
>
> so you have the created time, the progress time , the answered time and the
> hangup time
>
>
> you always have created time and hangup time but not always progress and
> answered
>
> so talk time is (hangup_time - (answered_time unless its 0)) or if
> answered_time is 0 there is no talk time
> ring time is:
>          (answered time unless its zero otherwise hangup_time) -
> created_time
>
>
> edit conf/autoload_configs/cdr_csv.conf.xml
>
> uncomment:
>
> <param name="debug" value="true"/>
>
> and restart FS
>
>
> now when you hangup all your calls you will see the values you have to work
> with in the cdr
>
> if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
>> way?
>>
>> I've realised since my last email that even having the progress_time
>> doesn't help without the progress_media_time or some way to find when the
>> ringing stopped - getting the server timestamp wouldn't happen until the
>> call ended, at which point I can't split the ring time from the talk time.
>>
>> Surely there can't be something that Asterisk handles well that FreeSwitch
>> doesn't? ;-)
>>
>> I'm looking into the CDR now - might have to parse the log files to get
>> the end time from there :-(
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> progress_media_time is only set if you get a 183
>>> if you get a 180 with no media it uses progress_time instead
>>>
>>> as long as you have one or the other you have the timestamp of when it
>>> started ringing.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Yeah, that's what I had expected, and was why I was confused.
>>>>
>>>> I've done some more tests, and tracked the results a little closer in a
>>>> spreadsheet, and it seems that when I call the extension directly from
>>>> another sipphone the  Caller-Channel-Answered-Time minus
>>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>>> fine.
>>>>
>>>> When I call into an IVR with the javascript dialplan and then create the
>>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>>> that are reported after the call ends are reported on the A-leg's Created &
>>>> Answered.
>>>>
>>>> The good news is that the Progress-Time is reported on when the B-leg
>>>> started ringing, so I can know when the call started ringing.
>>>> The bad news is that the Progress-Media-Time is always blank
>>>>
>>>> I can take the Progress-Time and compare it to the system-clock, and
>>>> that should generally be accurate to a second or so, which is better than
>>>> nothing, but it'd be nice to do it properly.
>>>>
>>>> Any other ideas? Is there any channel-variables I can set that would be
>>>> worth playing with? I'm currently only setting:
>>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> even better you have the progress and progress_media timestamps too
>>>>> so you can measure from the instance you got the first ringing
>>>>> indication
>>>>> so then you can also measure how long it took to start ringing too
>>>>> (PDD)
>>>>>
>>>>>
>>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>>
>>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>>> everything, except for tracking one important metric: How long the phone
>>>>>> rang before an agent picked it up.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 13:25:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 13:25:06 -0700
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
	<8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>
Message-ID: <87f2f3b91003171325q22cc746dw58675d0cb49b10d2@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:51 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Adjust the dial plan to call extension  8000. The default Dial plan like
> below
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> change to
>
> <condition field="destination_number" expression="^(*[1-8]*0[01][0-9])$">
>

Careful! This causes LOTS of different dialed numbers to match. You might
try this instead:
<condition field="destination_number" expression="^(8000|10[01][0-9])$">

This subject is covered nicely in the newbie article:
http://bit.ly/EpVrv

-MC
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From infos at madovsky.org  Wed Mar 17 13:26:05 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 16:26:05 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
	<191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>
Message-ID: <14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>

Sorry Anthony I never made an official update in C language with diff patch
so the change is at the line 620, I commented out the original line

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 3:55 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  could you get a svn diff of the switch_utils.c so we can  see what the change was?




  On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

    Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
    it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

    I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

    Regards

    Franck
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 16, 2010 4:35 PM
      Subject: Re: [Freeswitch-users] voicemail options


      ok, I tried 50 times to leave a message,
      it worked only one time.
      It seems that sendmail needs to be free to send the email....

        ----- Original Message ----- 
        From: devel at thom.fr.eu.org 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 16, 2010 3:59 PM
        Subject: Re: [Freeswitch-users] voicemail options


        Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

        I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



        Fran?ois





        De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
        Envoy? : mardi 16 mars 2010 20:30
        ? : freeswitch-users at lists.freeswitch.org
        Objet : Re: [Freeswitch-users] voicemail options



        are you running FS as root or a regular user?
        maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



        On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

        I noticed that sendmail makes a segfault when FS is trying to send email

        and sendmail has some emails locked in folder queue....



        Regards



        Franck


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




        -- 
        Anthony Minessale II

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

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

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



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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From msc at freeswitch.org  Wed Mar 17 13:34:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 13:34:16 -0700
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
Message-ID: <87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>

On Wed, Mar 17, 2010 at 1:10 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks, I suspect this is what I really needed:
>
>
> > if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
> I guess there's more for me to un-learn from Asterisk than I'd thought :-(
>

Yes, there probably is. FreeSWITCH is definitely a paradigm shift, but once
you realize the power under the hood you'll never go back to Asterisk.

If you *really* want to get fancy you can use mod_xml_cdr and get a true 3D
picture of your call. You can even use it to post the XML CDRs to an
external web server and do all your billing offline. :D

You might want to join #freeswitch on irc.freenode.net and rub elbows with
the other Asterisk refugees who've made the transition.

-MC
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From anthony.minessale at gmail.com  Wed Mar 17 13:37:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 15:37:57 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
	<191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>
	<14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
Message-ID: <191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>

FYI if you made the change in your working copy you could generate a patch
with

svn diff > patch.txt

After looking at your change, the problem is that many people depend on this
working the way it currently is.
Would you consider making a script that mimics the current behavior of a
mailer app and read the file from stdin and deliver the subsequent file so
it is backwards-compatible with the way everything already works?


On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

>  Sorry Anthony I never made an official update in C language with diff
> patch
> so the change is at the line 620, I commented out the original line
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 17, 2010 3:55 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> could you get a svn diff of the switch_utils.c so we can  see what the
> change was?
>
>
>
> On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Finally I succeed to send voicemail with sendmail through a little php
>> script (could be Perl also)
>> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
>> doesn't exist.
>>
>> I attached my php script and switch_utils.c. Not sure if the FS developers
>> will accept my modification
>>
>> Regards
>>
>> Franck
>>
>> ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 16, 2010 4:35 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>> ok, I tried 50 times to leave a message,
>> it worked only one time.
>> It seems that sendmail needs to be free to send the email....
>>
>>
>> ----- Original Message -----
>> *From:* devel at thom.fr.eu.org
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 16, 2010 3:59 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>>  Are you running debian, I have the same problem on debian x64 (already
>> mentioned here
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
>> and cannot fix it.
>>
>> I don?t know if it?s a priviledge problem, as I got the same results
>> running FS with or without ?u.
>>
>>
>>
>> Fran?ois
>>
>>
>>
>>
>>
>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
>> Minessale
>> *Envoy? :* mardi 16 mars 2010 20:30
>> *? :* freeswitch-users at lists.freeswitch.org
>> *Objet :* Re: [Freeswitch-users] voicemail options
>>
>>
>>
>> are you running FS as root or a regular user?
>> maybe there is a permission or resource limitation problem preventing it
>> from getting the larger stack space necessary?
>>
>>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> I noticed that sendmail makes a segfault when FS is trying to send email
>>
>> and sendmail has some emails locked in folder queue....
>>
>>
>>
>> Regards
>>
>>
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Wed Mar 17 13:55:55 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 16:55:55 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <1BB02984444E477196CA8761047E79C9@MOBILEE1705>

ok thanks for your help. I will do it next time.
since 3 days I tried hundred times with bash, mailx, mail, same result, somtimes it works sometimes segmnentation fault
with php way it works 100% perfectly now...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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




            -- 
            Anthony Minessale II

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

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

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

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
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  -- 
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  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
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From fraserredmond at gmail.com  Wed Mar 17 15:23:12 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 22:23:12 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com> 
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com> 
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
Message-ID: <c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>  If you *really* want to get fancy you can use mod_xml_cdr and get a true
> 3D picture of your call. You can even use it to post the XML CDRs to an
> external web server and do all your billing offline. :D
>


Yes, I was planning on using mod_xml_cdr to pass the info to our webserver
to update the database, once I know that mod_cdr_csv is getting the right
data.... which it still isn't  :-(

Much the same problem as before - in mod_cdr_csv's logs the start-time and
answered-time aren't what I expect. I'm gradually getting a better handle on
whats happening...

The 4 significant events in my dialplan are:

ivr-started  (I dial a switchboard number, and enter the extn I want to be
connected to)
call-started  (I'm connected to that extn)
call-ended  (the call with the extn ends)
ivr-ended  (I may return to the extn, or end at the same time as the call)

Master.csv contains 2 rows, both with from and to numbers matching the
caller and callee appropriately.

It seems like the first/blind_transfer row is not picking up the info from
the call that it ought to, and some of it is being recorded against the
second/main ivr row and then lost when it gets new data.

First row has:
start = call-started
answer = empty
end = call-started (probably actually call-connected)
hangup_cause = BLIND_TRANSFER
bleg uuid = empty

Second row (also appears in the separate csv file for the caller's
extension) has
start = ivr-started
answer = ivr-started
end = ivr-ended
hangup_cause = NORMAL_CLEARING
bleg uuid = present

Cheers,
Fraser
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From anthony.minessale at gmail.com  Wed Mar 17 15:37:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 17:37:56 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
	<c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>
Message-ID: <191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>

you might want to turn on B-leg logging

On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>  If you *really* want to get fancy you can use mod_xml_cdr and get a true
>> 3D picture of your call. You can even use it to post the XML CDRs to an
>> external web server and do all your billing offline. :D
>>
>
>
> Yes, I was planning on using mod_xml_cdr to pass the info to our webserver
> to update the database, once I know that mod_cdr_csv is getting the right
> data.... which it still isn't  :-(
>
> Much the same problem as before - in mod_cdr_csv's logs the start-time and
> answered-time aren't what I expect. I'm gradually getting a better handle on
> whats happening...
>
> The 4 significant events in my dialplan are:
>
> ivr-started  (I dial a switchboard number, and enter the extn I want to be
> connected to)
> call-started  (I'm connected to that extn)
> call-ended  (the call with the extn ends)
> ivr-ended  (I may return to the extn, or end at the same time as the call)
>
> Master.csv contains 2 rows, both with from and to numbers matching the
> caller and callee appropriately.
>
> It seems like the first/blind_transfer row is not picking up the info from
> the call that it ought to, and some of it is being recorded against the
> second/main ivr row and then lost when it gets new data.
>
> First row has:
> start = call-started
> answer = empty
> end = call-started (probably actually call-connected)
> hangup_cause = BLIND_TRANSFER
> bleg uuid = empty
>
> Second row (also appears in the separate csv file for the caller's
> extension) has
> start = ivr-started
> answer = ivr-started
> end = ivr-ended
> hangup_cause = NORMAL_CLEARING
> bleg uuid = present
>
> Cheers,
> Fraser
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From fraserredmond at gmail.com  Wed Mar 17 16:11:31 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 23:11:31 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com> 
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com> 
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com> 
	<c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com> 
	<191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>
Message-ID: <c32ee1cf1003171611i3c40fb2cte27b84ec920d7016@mail.gmail.com>

lol - yeah, I guess that would help!  /smacks head against nearby wall/

I'm in the UK and am heading to bed now - I'll sleep much better thanks to
those 8 words from you :-)

Thanks a bunch!

Cheers,
Fraser


On Wed, Mar 17, 2010 at 10:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you might want to turn on B-leg logging
>
> On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>>
>> It seems like the first/blind_transfer row is not picking up the info from
>> the call that it ought to, and some of it is being recorded against the
>> second/main ivr row and then lost when it gets new data.
>>
>
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From jmesquita at freeswitch.org  Wed Mar 17 16:37:01 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 20:37:01 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003171251.57460.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003170049.06565.robert.dyck@shaw.ca>
	<ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
	<201003171251.57460.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>

No, not at all. Do a lsof -i or a netstat. You should be able to see it all
there. Oh! Wait! Have you made a "make install" on freeswitch trunk?

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> I can see a process for fscomm. Should I be seeing a process for the core.
>
> When starting fscomm it reports -
> Initializing globals...
>
> Initializing core...
>
> Everything OK, Entering runtime loop ...
>
> BTW, not Windows but Fedora 11, KDE flavour
>
> Thanks for looking into this
> RD
>
>
> On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > Rob, I believe that the core has failed to load. I have not added any
> type
> > of check on the code level for those cases yet as I believed it would
> never
> > really fail to load. Those cases have been more frequent than what I
> > originally expected so I will add that to code today (hopefully later
> > tonight).
> >
> > Meanwhile, what you can do is check if the ports are open or not on the
> OS.
> > I am not sure how to do that on Windows. Or, maybe we already have logs
> on
> > the log folder?
> >
> > Regards,
> > Jo?o Mesquita
> >
> > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > Is there something wrong with my build?
> > >
> > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > > [Socket
> > > Connection Error]" with fscomm running.
> > >
> > > RD
> > >
> > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > With FSComm running you can use fs_cli to try pa list. If u can't
> list
> > >
> > > the
> > >
> > > > devices that would be the problem. I'm going to try to dig into this
> > > > further on 1 of my desktops w/ pulseaudio.
> > > >
> > > > -- W
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From infos at madovsky.org  Wed Mar 17 18:26:33 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 21:26:33 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <4F6DDBC6F1C04B45A6B70316C8D9FC66@MOBILEE1705>


>Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent >file so it is backwards-compatible with the way everything already works?

yes I can do it...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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            Anthony Minessale II

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

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
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      -- 
      Anthony Minessale II

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

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
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  -- 
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  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
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From robert.dyck at shaw.ca  Wed Mar 17 18:46:17 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 18:46:17 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171251.57460.robert.dyck@shaw.ca>
	<ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
Message-ID: <201003171846.19857.robert.dyck@shaw.ca>

I couldn't remember if I had done an install so I ran it. It didn't make any 
difference though.

Some observations
Freeswitch is not installed in my usual path
Fscomm is not installed by make install
Fs_cli only works if I start freeswitch first
Pa list ( as you suggested earlier ) is not a valid command

Here is the output of lsof -i
fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
>192.168.1.99:nat-pmp 
fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk 
fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk (LISTEN)
fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP localhost.localdomain:oa-
system (LISTEN)

I could not find anything in the netstat output that pertained to freeswitch or 
fscomm.


On Wednesday 17 March 2010 04:37:01 pm Jo?o Mesquita wrote:
> No, not at all. Do a lsof -i or a netstat. You should be able to see it all
> there. Oh! Wait! Have you made a "make install" on freeswitch trunk?
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > I can see a process for fscomm. Should I be seeing a process for the
> > core.
> > 
> > When starting fscomm it reports -
> > Initializing globals...
> > 
> > Initializing core...
> > 
> > Everything OK, Entering runtime loop ...
> > 
> > BTW, not Windows but Fedora 11, KDE flavour
> > 
> > Thanks for looking into this
> > RD
> > 
> > On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > > Rob, I believe that the core has failed to load. I have not added any
> > 
> > type
> > 
> > > of check on the code level for those cases yet as I believed it would
> > 
> > never
> > 
> > > really fail to load. Those cases have been more frequent than what I
> > > originally expected so I will add that to code today (hopefully later
> > > tonight).
> > > 
> > > Meanwhile, what you can do is check if the ports are open or not on the
> > 
> > OS.
> > 
> > > I am not sure how to do that on Windows. Or, maybe we already have logs
> > 
> > on
> > 
> > > the log folder?
> > > 
> > > Regards,
> > > Jo?o Mesquita
> > > 
> > > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > > Is there something wrong with my build?
> > > > 
> > > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > > > [Socket
> > > > Connection Error]" with fscomm running.
> > > > 
> > > > RD
> > > > 
> > > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > > With FSComm running you can use fs_cli to try pa list. If u can't
> > 
> > list
> > 
> > > > the
> > > > 
> > > > > devices that would be the problem. I'm going to try to dig into
> > > > > this further on 1 of my desktops w/ pulseaudio.
> > > > > 
> > > > > -- W
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > > > http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From infos at madovsky.org  Wed Mar 17 19:49:37 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 22:49:37 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <2D98E43C1BA344E892A250B33927991E@MOBILEE1705>

After dozen of test to correct my php script to be compatible without my patch I noticed
that the real problem comes from "cat" command and not sendmail. I don't know
why but sometimes (very often) the cat  can't write and make a broken pipe, that
of course make sendmail a segmentation fault. So for now I'm locked to use
the patched switch_utils.
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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            -- 
            Anthony Minessale II

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

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

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



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


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            http://www.freeswitch.org


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





      -- 
      Anthony Minessale II

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

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

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



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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From cucku.cucku at yahoo.com.vn  Wed Mar 17 19:51:45 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 18 Mar 2010 10:51:45 +0800 (SGT)
Subject: [Freeswitch-users] [roblem with reloadxml command
Message-ID: <585528.94129.qm@web76210.mail.sg1.yahoo.com>

Hi List

i am new to freeswitch, i do the reloadxml in Freeswitch console and get the error:

freeswitch at internal> reloadxml
2010-03-17 15:55:22.694226 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/internal/*.xml (No such file or directory)
+OK [Success]


Please help
Ha`



      __________________________________________________________
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From brian at freeswitch.org  Wed Mar 17 19:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 21:56:56 -0500
Subject: [Freeswitch-users] [roblem with reloadxml command
In-Reply-To: <585528.94129.qm@web76210.mail.sg1.yahoo.com>
References: <585528.94129.qm@web76210.mail.sg1.yahoo.com>
Message-ID: <0E28DEDC-90BD-414A-A56A-46DFF0060743@freeswitch.org>

You can ignore this.  It means you have nothing to include into the config in that directory.

/b

On Mar 17, 2010, at 9:51 PM, false wrote:

> Hi List
> 
> i am new to freeswitch, i do the reloadxml in Freeswitch console and get the error:
> 
> freeswitch at internal> reloadxml
> 2010-03-17 15:55:22.694226 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/internal/*.xml (No such file or directory)
> +OK [Success]
> 
> 
> Please help
> Ha`
> 

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From gkuri at ieee.org  Wed Mar 17 20:47:20 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 17 Mar 2010 20:47:20 -0700
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support [Resolved]
Message-ID: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>

We resolved the issue with Broadsoft SCA and inbound phone calls as it
relates to this thread.

If you're using a domain name rather than IP address for $${domain} in
vars.conf, you need to set the sip_invite_domain on the bridge app in
the dialplan, otherwise the INVITEs from FreeSWITCH end up being from
the IP address, which screws up SCA.

For example, we set sip_invite_domain=$${domain} on the bridge app for
local extensions in the dialplan ...

     <action application="bridge"
data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}"/>

Shouldn't this be part of the default config in SVN ?

Cheers,
Gabe


On Mon, Mar 1, 2010 at 11:01 PM, Brian West <brian at freeswitch.org> wrote:
> You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen.
>
> Verify the subscribe packet...
>
> /b
>
> On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:
>
>> I've torn my hair out for hours and I still can't get it to work with
>> the domain name set everywhere <sigh> ...
>>
>> I can get it to work if I hardcode the IP address on the phones and in
>> all the config files, but I can't get the domain name to work on
>> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
>> I'd think it wouldn't work on outgoing calls either, but that seems to
>> work fine.
>>
>> Any other ideas? Might it have anything to do with the incoming call
>> coming in from the PSTN and going through the external profile?
>>
>> Cheers,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jmesquita at freeswitch.org  Wed Mar 17 21:17:51 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 18 Mar 2010 01:17:51 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003171846.19857.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171251.57460.robert.dyck@shaw.ca>
	<ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
	<201003171846.19857.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>

I think that the problem is that the core is not able to find the
modules....

Try installing the mods on /usr/local/freeswitch/mods. My guess is that
everything will load just fine.

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 10:46 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> I couldn't remember if I had done an install so I ran it. It didn't make
> any
> difference though.
>
> Some observations
> Freeswitch is not installed in my usual path
> Fscomm is not installed by make install
> Fs_cli only works if I start freeswitch first
> Pa list ( as you suggested earlier ) is not a valid command
>
> Here is the output of lsof -i
> fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
> >192.168.1.99:nat-pmp
> fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk
> fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk
> (LISTEN)
> fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP
> localhost.localdomain:oa-
> system (LISTEN)
>
> I could not find anything in the netstat output that pertained to
> freeswitch or
> fscomm.
>
>
> On Wednesday 17 March 2010 04:37:01 pm Jo?o Mesquita wrote:
> > No, not at all. Do a lsof -i or a netstat. You should be able to see it
> all
> > there. Oh! Wait! Have you made a "make install" on freeswitch trunk?
> >
> > Regards,
> > Jo?o Mesquita
> >
> > On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > I can see a process for fscomm. Should I be seeing a process for the
> > > core.
> > >
> > > When starting fscomm it reports -
> > > Initializing globals...
> > >
> > > Initializing core...
> > >
> > > Everything OK, Entering runtime loop ...
> > >
> > > BTW, not Windows but Fedora 11, KDE flavour
> > >
> > > Thanks for looking into this
> > > RD
> > >
> > > On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > > > Rob, I believe that the core has failed to load. I have not added any
> > >
> > > type
> > >
> > > > of check on the code level for those cases yet as I believed it would
> > >
> > > never
> > >
> > > > really fail to load. Those cases have been more frequent than what I
> > > > originally expected so I will add that to code today (hopefully later
> > > > tonight).
> > > >
> > > > Meanwhile, what you can do is check if the ports are open or not on
> the
> > >
> > > OS.
> > >
> > > > I am not sure how to do that on Windows. Or, maybe we already have
> logs
> > >
> > > on
> > >
> > > > the log folder?
> > > >
> > > > Regards,
> > > > Jo?o Mesquita
> > > >
> > > > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca>
> wrote:
> > > > > Is there something wrong with my build?
> > > > >
> > > > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error
> Connecting
> > > > > [Socket
> > > > > Connection Error]" with fscomm running.
> > > > >
> > > > > RD
> > > > >
> > > > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > > > With FSComm running you can use fs_cli to try pa list. If u can't
> > >
> > > list
> > >
> > > > > the
> > > > >
> > > > > > devices that would be the problem. I'm going to try to dig into
> > > > > > this further on 1 of my desktops w/ pulseaudio.
> > > > > >
> > > > > > -- W
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yehavi.bourvine at gmail.com  Wed Mar 17 22:03:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 18 Mar 2010 07:03:26 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support [Resolved]
In-Reply-To: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>
References: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>
Message-ID: <d361ca7b1003172203x587215fcr9c7bb4a05f51698b@mail.gmail.com>

Thanks! You've done my day...

I am using IPs in order to make SCA working, but this prevents Polycom to do
TLS (it need names). Now both can be made working.

                     Thanks again, __Yehavi:

2010/3/18 Gabriel Kuri <gkuri at ieee.org>

> We resolved the issue with Broadsoft SCA and inbound phone calls as it
> relates to this thread.
>
> If you're using a domain name rather than IP address for $${domain} in
> vars.conf, you need to set the sip_invite_domain on the bridge app in
> the dialplan, otherwise the INVITEs from FreeSWITCH end up being from
> the IP address, which screws up SCA.
>
> For example, we set sip_invite_domain=$${domain} on the bridge app for
> local extensions in the dialplan ...
>
>     <action application="bridge"
> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
> ${domain_name}"/>
>
> Shouldn't this be part of the default config in SVN ?
>
> Cheers,
> Gabe
>
>
> On Mon, Mar 1, 2010 at 11:01 PM, Brian West <brian at freeswitch.org> wrote:
> > You have to watch phones that might do the switch on you... like resolve
> the IP then do the subscribe incorrectly... that could happen.
> >
> > Verify the subscribe packet...
> >
> > /b
> >
> > On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:
> >
> >> I've torn my hair out for hours and I still can't get it to work with
> >> the domain name set everywhere <sigh> ...
> >>
> >> I can get it to work if I hardcode the IP address on the phones and in
> >> all the config files, but I can't get the domain name to work on
> >> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
> >> I'd think it wouldn't work on outgoing calls either, but that seems to
> >> work fine.
> >>
> >> Any other ideas? Might it have anything to do with the incoming call
> >> coming in from the PSTN and going through the external profile?
> >>
> >> Cheers,
> >> Gabe
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Mar 17 22:37:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:37:33 -0700
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
Message-ID: <87f2f3b91003172237l57351ae1g53ca346ce150f6f5@mail.gmail.com>

FYI this is now on the wiki:
http://wiki.freeswitch.org/wiki/CDR#Calculating_Various_Time_Values_For_A_Call

-MC

On Wed, Mar 17, 2010 at 12:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i'll try again
>
> progress_media and progress are both the same for your purposes
> you will only have exactly one of the 2 set.  its the exact time it started
> ringing.
>
> for our sake, if we have either of them we will take the one that is not
> zero and call it the progress time.
>
> so you have the created time, the progress time , the answered time and the
> hangup time
>
>
> you always have created time and hangup time but not always progress and
> answered
>
> so talk time is (hangup_time - (answered_time unless its 0)) or if
> answered_time is 0 there is no talk time
> ring time is:
>          (answered time unless its zero otherwise hangup_time) -
> created_time
>
>
> edit conf/autoload_configs/cdr_csv.conf.xml
>
> uncomment:
>
> <param name="debug" value="true"/>
>
> and restart FS
>
>
> now when you hangup all your calls you will see the values you have to work
> with in the cdr
>
> if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
>> way?
>>
>> I've realised since my last email that even having the progress_time
>> doesn't help without the progress_media_time or some way to find when the
>> ringing stopped - getting the server timestamp wouldn't happen until the
>> call ended, at which point I can't split the ring time from the talk time.
>>
>> Surely there can't be something that Asterisk handles well that FreeSwitch
>> doesn't? ;-)
>>
>> I'm looking into the CDR now - might have to parse the log files to get
>> the end time from there :-(
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> progress_media_time is only set if you get a 183
>>> if you get a 180 with no media it uses progress_time instead
>>>
>>> as long as you have one or the other you have the timestamp of when it
>>> started ringing.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Yeah, that's what I had expected, and was why I was confused.
>>>>
>>>> I've done some more tests, and tracked the results a little closer in a
>>>> spreadsheet, and it seems that when I call the extension directly from
>>>> another sipphone the  Caller-Channel-Answered-Time minus
>>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>>> fine.
>>>>
>>>> When I call into an IVR with the javascript dialplan and then create the
>>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>>> that are reported after the call ends are reported on the A-leg's Created &
>>>> Answered.
>>>>
>>>> The good news is that the Progress-Time is reported on when the B-leg
>>>> started ringing, so I can know when the call started ringing.
>>>> The bad news is that the Progress-Media-Time is always blank
>>>>
>>>> I can take the Progress-Time and compare it to the system-clock, and
>>>> that should generally be accurate to a second or so, which is better than
>>>> nothing, but it'd be nice to do it properly.
>>>>
>>>> Any other ideas? Is there any channel-variables I can set that would be
>>>> worth playing with? I'm currently only setting:
>>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> even better you have the progress and progress_media timestamps too
>>>>> so you can measure from the instance you got the first ringing
>>>>> indication
>>>>> so then you can also measure how long it took to start ringing too
>>>>> (PDD)
>>>>>
>>>>>
>>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>>
>>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>>> everything, except for tracking one important metric: How long the phone
>>>>>> rang before an agent picked it up.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 22:46:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:46:47 -0700
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
	<2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
Message-ID: <87f2f3b91003172246u4d8b81a1r29d7aef0dd0da51@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:05 AM, Madovsky <infos at madovsky.org> wrote:

> So why not GPL or Gnu Affero public licence ?
>
> Long story short: the FS devs wanted a different kind of freedom than that
which is offered by the GPL. The GPL is "viral" in that it forces everything
it touches to by GPL-compatible. This kind of force-fed freedom doesn't
always fit well with software that is technically a library (libfreeswitch)
that can be compiled/embedded in other projects.

Bottom line: no OSS license can be all things to all projects. Tony chose
the best one available for his specific project.
-MC
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From msc at freeswitch.org  Wed Mar 17 22:48:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:48:41 -0700
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
	<6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
Message-ID: <87f2f3b91003172248t33e5ff00nd95dc1e2d8c826c3@mail.gmail.com>

Evidently Google doesn't have 40 million Gizmo users complaining about how
they broke their INVITEs...
-MC

On Wed, Mar 17, 2010 at 10:15 AM, Brian West <brian at freeswitch.org> wrote:

> Sure... I would love that.
>
> /b
>
> On Mar 17, 2010, at 12:11 PM, Madovsky wrote:
>
> But Brian,
>
> are you ready to answer to 40 millions users if  the bug is not resolved ?
> :D
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 23:02:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 23:02:17 -0700
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
	<8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
Message-ID: <87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>

On Wed, Mar 17, 2010 at 11:54 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

>  <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
> mber}"/>
>
> I setup the above dial plan. The call to extension works perfect. But It
> never call to the ${followme_neumber}.
>
> How can I fix this issue.
>
> Thanks
> Lloyd
>
> You might need to set this variable to true prior to your bridge:
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_continue_on_timeout

-MC
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From infos at madovsky.org  Wed Mar 17 23:02:18 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 02:02:18 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705><6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
	<87f2f3b91003172248t33e5ff00nd95dc1e2d8c826c3@mail.gmail.com>
Message-ID: <F819F014EE9B42D29D4837EDA1189C26@MOBILEE1705>

maybe it's to technical to make users understand ;)
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 1:48 AM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Evidently Google doesn't have 40 million Gizmo users complaining about how they broke their INVITEs...
  -MC


  On Wed, Mar 17, 2010 at 10:15 AM, Brian West <brian at freeswitch.org> wrote:

    Sure... I would love that.


    /b


    On Mar 17, 2010, at 12:11 PM, Madovsky wrote:


      But Brian,

      are you ready to answer to 40 millions users if  the bug is not resolved ? :D





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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From xengelpublicx at gmail.com  Wed Mar 17 23:11:33 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 09:11:33 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
Message-ID: <eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>

I need to make a queue for the Secretary, in which one secretary. But
if the Secretary is not there, I would like to be able to intercept
calls to other users. This can not be done even if the intercept will
be a user to the agent queue?

On Wed, Mar 17, 2010 at 10:11 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Mar 17, 2010 at 5:22 AM, Vladimir Elizarov <xengelpublicx at gmail.com>
> wrote:
>>
>> Yes
>>
>
> You can't. The FIFO system isn't transferring the call to the agent's phone.
> Rather, it is ringing the agent's phone and when the agent picks up, it then
> bridges the caller to the agent.
>
> What exactly are you trying to accomplish? Just curious. By the way you can
> possibly resolve this issue by create an "agent access" extension. The idea
> is simple: User A picks up and dials an extension; that extension then
> connects User A to the longest waiting caller in the FIFO queue.

I need to make a queue for the Secretary. In this stage only one
secretary. But if the Secretary is not there, I would like to be able
to intercept input calls in fifo queue to other users. This can not be
done even if the intercept will be a user to the agent queue?
>
> Use this example to create your "agent access" extension:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Take_a_caller_out_of_a_FIFO_queue
>
> The net result is probably the same: a third phone connects to the caller in
> queue. If that caller is the only one in the queue then the agent's phone
> that is ringing will stop ringing immediately.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov


From msc at freeswitch.org  Wed Mar 17 23:59:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 23:59:42 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
Message-ID: <87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>

On Wed, Mar 17, 2010 at 11:11 PM, Vladimir Elizarov <xengelpublicx at gmail.com
> wrote:

> I need to make a queue for the Secretary, in which one secretary. But
> if the Secretary is not there, I would like to be able to intercept
> calls to other users. This can not be done even if the intercept will
> be a user to the agent queue?
>
> Why not just have the receptionist log out of the queue when she's not at
her desk?
-MC
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From robert.dyck at shaw.ca  Thu Mar 18 00:00:05 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Thu, 18 Mar 2010 00:00:05 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171846.19857.robert.dyck@shaw.ca>
	<ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
Message-ID: <201003180000.07647.robert.dyck@shaw.ca>

After figuring out how to get portaudio to build and install, I thought I would 
get somewhere. Alas, it was not to be.

Attempting to load mod_portaudio put me back where I started - no devices

2010-03-17 23:51:02.007720 [ERR] mod_portaudio.c:980 Cannot find an input 
device
2010-03-17 23:51:02.009147 [ERR] mod_portaudio.c:990 Cannot find an input 
device                                                                        
2010-03-17 23:51:02.009147 [CRIT] switch_loadable_module.c:882 Error Loading 
module /usr/local/freeswitch/mod/mod_portaudio.so                          
**Module load routine returned an error** 

Probably, I can't test tomorrow. It's my day with the grandchildren.

RD

On Wednesday 17 March 2010 09:17:51 pm Jo?o Mesquita wrote:
> I think that the problem is that the core is not able to find the
> modules....
> 
> Try installing the mods on /usr/local/freeswitch/mods. My guess is that
> everything will load just fine.
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 10:46 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > I couldn't remember if I had done an install so I ran it. It didn't make
> > any
> > difference though.
> > 
> > Some observations
> > Freeswitch is not installed in my usual path
> > Fscomm is not installed by make install
> > Fs_cli only works if I start freeswitch first
> > Pa list ( as you suggested earlier ) is not a valid command
> > 
> > Here is the output of lsof -i
> > fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
> > 
> > >192.168.1.99:nat-pmp
> > 
> > fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk
> > fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk
> > (LISTEN)
> > fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP
> > localhost.localdomain:oa-
> > system (LISTEN)
> > 
> > I could not find anything in the netstat output that pertained to
> > freeswitch or
> > fscomm.


From tculjaga at gmail.com  Thu Mar 18 00:08:04 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 18 Mar 2010 08:08:04 +0100
Subject: [Freeswitch-users] Release Cause in response message
In-Reply-To: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>
References: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>
Message-ID: <65d96fc81003180008l2238f9f9vac5545ce605787e5@mail.gmail.com>

any way to pass a disconnect reason vi a respond command ???

anyone?

T.

On Wed, Mar 17, 2010 at 3:46 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello folks,
>
>
> anyone knows how to pass a Release cause/ Disconnect Reason when firing
> respond from a dialplan ... e.g.:
>
> Ive tried with sip_h_Reason=Q.850;cause=27 but it is not working :)
>
>
>
>    <extension name="parseResult">
>       <condition field="destination_number" expression="^parseResult$"/>
>       <condition field="${componentStatus}" expression="^1$"
> break="on-true">
>          <action application="log" data="INFO ######################## CALL
> FAILED REASON 1' ##########\n"/>
>
>          <action application="set" data="sip_h_Reason=Q.850;cause=16"/>
>
>          <action application="respond" data="480 Temporarily Unavailable"/>
>       </condition>
>    </extension>
>
>
> any hint?
>
>
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From xengelpublicx at gmail.com  Thu Mar 18 00:39:59 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 10:39:59 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
Message-ID: <eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>

If she leaves the queue, the queue will not be agents. A user simply
intercepting the call should be able to transfer the call to your
phone when the Secretary is not there.

Agents can intercept a call in the fifo queue?

On Thu, Mar 18, 2010 at 9:59 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Mar 17, 2010 at 11:11 PM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>>
>> I need to make a queue for the Secretary, in which one secretary. But
>> if the Secretary is not there, I would like to be able to intercept
>> calls to other users. This can not be done even if the intercept will
>> be a user to the agent queue?
>>
> Why not just have the receptionist log out of the queue when she's not at
> her desk?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov


From brian at microcomaustralia.com.au  Thu Mar 18 00:56:49 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Mar 2010 18:56:49 +1100
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>
References: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
	<4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>
Message-ID: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>

On 17 March 2010 21:57, Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> Make the appropriate changes to the configuration files in /etc/dahdi/. Take a look at the files in /proc/dahdi/.

Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? I tried
copying the file, it doesn't seem to have changed anything...

huey:~# cat /proc/dahdi/1
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

	   1 WCTDM/4/0
	   2 WCTDM/4/1
	   3 WCTDM/4/2
	   4 WCTDM/4/3
huey:~# cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au


What process reads this file?
-- 
Brian May <brian at microcomaustralia.com.au>


From maciej.aniserowicz at gmail.com  Thu Mar 18 01:19:30 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 18 Mar 2010 00:19:30 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
Message-ID: <1268900370911-4755257.post@n2.nabble.com>


You're right, sorry.

I updated to trunk (rev 17031) and the problem is still here.

Logs produced by newest version: http://pastebin.freeswitch.org/12474. They
contain comments and commands that I used.

Scenario:
1st call is made and is parked. Then 2nd and 3rd calls are:
1) dialed
2) parked
3) bridged with 1st call
4) hanged up

It is OK for now.
Then a 4th call is dialed, parked, bridged and killed. After killing the 4th
call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.


I hope this description and logs help resolve the issue.
Please let me know if you need any more information.

FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cucku.cucku at yahoo.com.vn  Thu Mar 18 01:31:15 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 18 Mar 2010 16:31:15 +0800 (SGT)
Subject: [Freeswitch-users] Newbie question on domain name
Message-ID: <554663.90897.qm@web76203.mail.sg1.yahoo.com>

Hi List

i feel strange about default domain in Freswitch:
i read the vars.xml and default domain name is IP address:
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

and i register successfull the 1000, 1002 extension
the i copy 1000.xml to 7000.xml 
then i register 7000 extension but get error

Please help
Ha`

Registrations:
=================================================================================================
Call-ID:??????? 56dc9e359865070c at 192.168.1.6
User:?????????? 1002 at 192.168.1.11
Contact:??????? "user" <sip:1002 at 192.168.1.6:4318;user=phone>
Agent:????????? Grandstream BT110 1.0.7.3
Status:???????? Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
Host:?????????? fs
IP:???????????? 192.168.1.6
Port:?????????? 4318
Auth-User:????? 1002
Auth-Realm:???? 192.168.1.11
MWI-Account:??? 1002 at 192.168.1.11

Call-ID:??????? 8fa759179877592f at 192.168.1.12
User:?????????? 1009 at 192.168.1.11
Contact:??????? "user" <sip:1009 at 192.168.1.12:57386;user=phone>
Agent:????????? Grandstream BT110 1.0.7.3
Status:???????? Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
Host:?????????? fs
IP:???????????? 192.168.1.12
Port:?????????? 57386
Auth-User:????? 1009
Auth-Realm:???? 192.168.1.11
MWI-Account:??? 1009 at 192.168.1.11

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

freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866 Can't find user [7000 at 192.168.1.11]
You must define a domain called '192.168.1.11' in your directory and add a user with the id="7000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth failure (REGISTER) on sofia profile 'internal' for [7000 at 192.168.1.11] from ip 192.168.1.2




      Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n! T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?. http://vn.messenger.yahoo.com/pingbox/
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From tjardick at vanderkraan.net  Thu Mar 18 01:51:03 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 09:51:03 +0100
Subject: [Freeswitch-users] Newbie question on domain name
In-Reply-To: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
References: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
Message-ID: <B2EE99FE-EDE1-4717-BF06-536F522B1143@vanderkraan.net>

Hi False,

Did you edit the 7000.xml to reflect the changes within the file?

Also did you perform a "reloadxml" to tell freeswitch to reread it's  
configuration?


Regards,

Tj

On 18 Mar 2010, at 09:31, false wrote:

> Hi List
>
> i feel strange about default domain in Freswitch:
> i read the vars.xml and default domain name is IP address:
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
> and i register successfull the 1000, 1002 extension
> the i copy 1000.xml to 7000.xml
> then i register 7000 extension but get error
>
> Please help
> Ha`
>
> Registrations:
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> ======================================================================
> Call-ID:        56dc9e359865070c at 192.168.1.6
> User:           1002 at 192.168.1.11
> Contact:        "user" <sip:1002 at 192.168.1.6:4318;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
> Host:           fs
> IP:             192.168.1.6
> Port:           4318
> Auth-User:      1002
> Auth-Realm:     192.168.1.11
> MWI-Account:    1002 at 192.168.1.11
>
> Call-ID:        8fa759179877592f at 192.168.1.12
> User:           1009 at 192.168.1.11
> Contact:        "user" <sip:1009 at 192.168.1.12:57386;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
> Host:           fs
> IP:             192.168.1.12
> Port:           57386
> Auth-User:      1009
> Auth-Realm:     192.168.1.11
> MWI-Account:    1009 at 192.168.1.11
>
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> ======================================================================
>
> freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866  
> Can't find user [7000 at 192.168.1.11]
> You must define a domain called '192.168.1.11' in your directory and  
> add a user with the id="7000" attribute
> and you must configure your device to use the proper domain in it's  
> authentication credentials.
> 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth  
> failure (REGISTER) on sofia profile 'internal' for  
> [7000 at 192.168.1.11] from ip 192.168.1.2
>
>
> Truy c?p Mail nhanh h?n!
> Yahoo! khuy?n kh?ch b?n n?ng c?p tr?nh duy?t l?n Internet  
> Explorer 8 m?i, t?i ?u h?a cho Yahoo!. T?i t?i ??y!  
> (Mi?n ph?)_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Thu Mar 18 04:39:46 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Mar 2010 22:39:46 +1100
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003180000.07647.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171846.19857.robert.dyck@shaw.ca>
	<ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
	<201003180000.07647.robert.dyck@shaw.ca>
Message-ID: <20100318113946.GA1716@jdc.jasonjgw.net>

Rob Dyck <robert.dyck at shaw.ca> wrote:
> After figuring out how to get portaudio to build and install, I thought I
> would get somewhere. Alas, it was not to be.
> 
> Attempting to load mod_portaudio put me back where I started - no devices
> 
> 2010-03-17 23:51:02.007720 [ERR] mod_portaudio.c:980 Cannot find an input
> device 

I've seen this before; it was a permission problem, solved by adding the
freeswitch user to the audio group so that it could access the ALSA devices.

Whether this is the cause in your case depends on how your operating system
controls access to audio devices.



From rentmycoder at gmail.com  Thu Mar 18 05:21:09 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Thu, 18 Mar 2010 13:21:09 +0100
Subject: [Freeswitch-users] how to dial out through a local sip endpoint -
	mv372 mobile gateway adapter
Message-ID: <50e456911003180521je008a2dme94fcc6901a04b99@mail.gmail.com>

Hi guys,
Freeswitch rocks!
I've registered an Portech mv-372 mobile adapter to FS as an internal
user as userid/extension=5000.
How to dial out through this endpint?

In asterisk I've used Dial(sip/1234567 at 5000) where 1234567 is the
phone no to call...

In FS: bridge(sophia/internal/1234567 at 5000)
mod_dptools: normal_temporery_failure

any ideas how to do the bridge?
or should I setup the adapter as a gateway profile?

Just to make sure the adapter is ok i've tested it using asterisk and
it worked and was set up like this:
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk


From phunk0000 at hotmail.com  Thu Mar 18 05:22:10 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 08:22:10 -0400
Subject: [Freeswitch-users] FS Best practices
Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>

Hey List, still kinda new to this whole deal, looking for some advice.  We
are building a website that needs to interact with FS.  Not looking for
specific instructions just opinions about the best practices for doing
things like click to dial from a web page, calling CDR records to web
applications to display customers stats and general web integration.  Does
this all need to be done using scripting languages like Lua or Perl, or can
you directly integrate FS with a website in some other manner. Again, just
getting into all this, but having a blast learning FS, config and power
rock!  Anyhow, some places or things I should start looking into would be
greatly appreciated.  Thanks List!

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From gmaruzz at celliax.org  Thu Mar 18 05:33:13 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 18 Mar 2010 13:33:13 +0100
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
Message-ID: <7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.? We
> are building a website that needs to interact with FS.? Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.? Does
> this all need to be done using scripting languages like Lua or Perl, or can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.



> getting into all this, but having a blast learning FS, config and power
> rock!? Anyhow, some places or things I should start looking into would be
> greatly appreciated.? Thanks List!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steve at justfone.com  Thu Mar 18 05:39:10 2010
From: steve at justfone.com (Steven Brown)
Date: Thu, 18 Mar 2010 12:39:10 +0000
Subject: [Freeswitch-users] Re : how to dial out through a local sip
	endpoint - mv372 mobile > gateway adapter (rentmycoder rentmycoder)
Message-ID: <3e6d7b0c1003180539y5d8b300bl4392f72c34129ba3@mail.gmail.com>

Hi,

You can do it with the Portech  registered on FS as a regular endpoint
then  bridge to it
as below or similar

<extension name="gsm">
     <condition field="destination_number" expression="^9(.*)$">
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="bridge" data="sofia/internal/1005^$1@${domain}">

where the Portech registers as 1005 and I prefix the number I wish to
dial with 9

or alternatively just bridge directly to it without having to register

<action application="bridge" data="sofia/default/$1 at 192.168.1.3:5060"/>

Where 192.168.1.3 is the Portech

In both cases you need to configure the Portech Lan to Mobile Table as below

Item  URL                 Call Num
0       192.168.1.2     #

where 192.168.1.2 is my FS box

Steve


> Message: 5
> Date: Thu, 18 Mar 2010 13:21:09 +0100
> From: rentmycoder rentmycoder <rentmycoder at gmail.com>
> Subject: [Freeswitch-users] how to dial out through a local sip
> ? ? ? ?endpoint - ? ? ?mv372 mobile gateway adapter
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> ? ? ? ?<50e456911003180521je008a2dme94fcc6901a04b99 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi guys,
> Freeswitch rocks!
> I've registered an Portech mv-372 mobile adapter to FS as an internal
> user as userid/extension=5000.
> How to dial out through this endpint?
>
> In asterisk I've used Dial(sip/1234567 at 5000) where 1234567 is the
> phone no to call...
>
> In FS: bridge(sophia/internal/1234567 at 5000)
> mod_dptools: normal_temporery_failure
>
> any ideas how to do the bridge?
> or should I setup the adapter as a gateway profile?
>
> Just to make sure the adapter is ok i've tested it using asterisk and
> it worked and was set up like this:
> http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>
>
>
> ------------------------------
>
> Message: 6
> Date: Thu, 18 Mar 2010 08:22:10 -0400
> From: "Todd" <phunk0000 at hotmail.com>
> Subject: [Freeswitch-users] FS Best practices
> To: "FreeSWITCH-users at lists. freeswitch. org "
> ? ? ? ?<FreeSWITCH-users at lists.freeswitch.org>
> Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0 at phx.gbl>
> Content-Type: text/plain; charset="us-ascii"
>
> Hey List, still kinda new to this whole deal, looking for some advice. ?We
> are building a website that needs to interact with FS. ?Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration. ?Does
> this all need to be done using scripting languages like Lua or Perl, or can
> you directly integrate FS with a website in some other manner. Again, just
> getting into all this, but having a blast learning FS, config and power
> rock! ?Anyhow, some places or things I should start looking into would be
> greatly appreciated. ?Thanks List!
>
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> End of FreeSWITCH-users Digest, Vol 45, Issue 225
> *************************************************
>



-- 
Steven Brown

email   steve at justfone.com
office   08707706968
mobile 07768755409
fax      07884636663

Justfone - Company Reg. No.  : 3926817

Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW

The contents of this e-mail may be privileged and are confidential. It
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nor copied in any way. If received in error, please advise sender,
then delete it from your system. Internet email communications are not
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the contents of this message. Any views or opinions presented are
solely those of the author and do not necessarily represent those of
Justfone unless otherwise specifically stated.


From Russell.Mosemann at cune.org  Thu Mar 18 06:23:19 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 18 Mar 2010 13:23:19 -0000
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
Message-ID: <20100318132319.2A5BA312875@cuneorg-email.cune.pri>

Brian May <brian at microcomaustralia.com.au> said:

> Is /etc/dahdi/system.conf the same as /etc/zaptel.conf?

Just glancing through the file, it looks the same.

> What process reads this file?

/etc/init.d/dahdi

What is in /etc/dahdi/modules?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From lloyd.aloysius at gmail.com  Thu Mar 18 06:32:05 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 09:32:05 -0400
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
	<8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
	<87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>
Message-ID: <8a19bf2e1003180632h3b453042q7860cc7852421191@mail.gmail.com>

Michael, Thank you. I will try your suggestion.

Thanks
Lloyd

On Thu, Mar 18, 2010 at 2:02 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Mar 17, 2010 at 11:54 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>>  <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
>> mber}"/>
>>
>> I setup the above dial plan. The call to extension works perfect. But It
>> never call to the ${followme_neumber}.
>>
>> How can I fix this issue.
>>
>> Thanks
>> Lloyd
>>
>> You might need to set this variable to true prior to your bridge:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_continue_on_timeout
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 18 06:55:37 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Mar 2010 06:55:37 -0700
Subject: [Freeswitch-users] Newbie question on domain name
In-Reply-To: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
References: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
Message-ID: <4712D923-82FF-45D8-B73C-A8CF3816E2AB@freeswitch.org>

Did you edit 7000.xml? Make sure that you open that file and put  
"7000" in place of "1000".
-MC
Sent from my iPhone

On Mar 18, 2010, at 1:31 AM, false <cucku.cucku at yahoo.com.vn> wrote:

> Hi List
>
> i feel strange about default domain in Freswitch:
> i read the vars.xml and default domain name is IP address:
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
> and i register successfull the 1000, 1002 extension
> the i copy 1000.xml to 7000.xml
> then i register 7000 extension but get error
>
> Please help
> Ha`
>
> Registrations:
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> ======================================================================
> Call-ID:        56dc9e359865070c at 192.168.1.6
> User:           1002 at 192.168.1.11
> Contact:        "user" <sip:1002 at 192.168.1.6:4318;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
> Host:           fs
> IP:             192.168.1.6
> Port:           4318
> Auth-User:      1002
> Auth-Realm:     192.168.1.11
> MWI-Account:    1002 at 192.168.1.11
>
> Call-ID:        8fa759179877592f at 192.168.1.12
> User:           1009 at 192.168.1.11
> Contact:        "user" <sip:1009 at 192.168.1.12:57386;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
> Host:           fs
> IP:             192.168.1.12
> Port:           57386
> Auth-User:      1009
> Auth-Realm:     192.168.1.11
> MWI-Account:    1009 at 192.168.1.11
>
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> ======================================================================
>
> freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866  
> Can't find user [7000 at 192.168.1.11]
> You must define a domain called '192.168.1.11' in your directory and  
> add a user with the id="7000" attribute
> and you must configure your device to use the proper domain in it's  
> authentication credentials.
> 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth  
> failure (REGISTER) on sofia profile 'internal' for  
> [7000 at 192.168.1.11] from ip 192.168.1.2
>
>
> Truy c?p Mail nhanh h?n!
> Yahoo! khuy?n kh?ch b?n n?ng c?p tr?nh duy?t l?n Internet  
> Explorer 8 m?i, t?i ?u h?a cho Yahoo!. T?i t?i ??y!  
> (Mi?n ph?)
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From moises.silva at gmail.com  Thu Mar 18 07:19:34 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 18 Mar 2010 10:19:34 -0400
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <20100318132319.2A5BA312875@cuneorg-email.cune.pri>
References: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
	<20100318132319.2A5BA312875@cuneorg-email.cune.pri>
Message-ID: <c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>

do svn update ( that will fix some of the errors but NOT the critical one).
The critical one is "failure configuring device ... " and I think that one
is because you did not run "dahdi_cfg" utility.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Thu, Mar 18, 2010 at 9:23 AM, <Russell.Mosemann at cune.org> wrote:

> Brian May <brian at microcomaustralia.com.au> said:
>
> > Is /etc/dahdi/system.conf the same as /etc/zaptel.conf?
>
> Just glancing through the file, it looks the same.
>
> > What process reads this file?
>
> /etc/init.d/dahdi
>
> What is in /etc/dahdi/modules?
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Thu Mar 18 07:38:01 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 10:38:01 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
Message-ID: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>

Hi All,

The following lua script for calculate the call duration. But the issue here
, the duration is calculating from the outbound initiate time.

How to get the call answer time?


session:preAnswer();
-- sleep a second
session:sleep(1000);
--
digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav",
"invalid-digits.wav", "\\d+|\\*");

-- Initiate an outbound call
outSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session)


starttime  = os.time();
freeswitch.consoleLog("info",   "Start Time"..starttime.." \n");

freeswitch.consoleLog("info",   "Before Bridge the Call \n");
freeswitch.bridge(session, outSession);
freeswitch.consoleLog("info",   "After Bridge the Call \n");

endtime = os.time();
freeswitch.consoleLog("info",   "End Time"..endtime.." \n");

billsec = os.difftime(endtime,starttime);
freeswitch.consoleLog("info",   "Blling Sec"..billsec.." \n");

-- hangup
session:hangup();


Thanks
Lloyd
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From msc at freeswitch.org  Thu Mar 18 08:16:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:16:37 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
Message-ID: <87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>

On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> The following lua script for calculate the call duration. But the issue
> here , the duration is calculating from the outbound initiate time.
>
> How to get the call answer time?
>
If you're trying to do billing in-line you might as well stop right now and
read the CDRs page on the wiki. Trying to calculate this from inside the
session is silly.
-MC
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From msc at freeswitch.org  Thu Mar 18 08:29:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:29:24 -0700
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
Message-ID: <87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> > Hey List, still kinda new to this whole deal, looking for some advice.
> We
> > are building a website that needs to interact with FS.  Not looking for
> > specific instructions just opinions about the best practices for doing
> > things like click to dial from a web page, calling CDR records to web
> > applications to display customers stats and general web integration.
> Does
> > this all need to be done using scripting languages like Lua or Perl, or
> can
> > you directly integrate FS with a website in some other manner. Again,
> just
>
> You definitely want to have a look at:
> http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
> pages.
>
>
I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 08:32:54 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 11:32:54 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
Message-ID: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>

Thank you for the suggestion. Yes I went to the direction then Stop. How to
handle Hangup actions for every call?

For Example : I do not want to calculate the duration for all the calls or
execute actions for all the calls. I need calculate for particular calls
only.I tested the following ,* it execute for every calls hangup*. How to
control or How to execute in some conditions.


cdr_csv.conf.xml


<param name="default-template" value="lua"/>

<template name="lua">${lua(onhangup.lua)}</template>


On Thu, Mar 18, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> The following lua script for calculate the call duration. But the issue
>> here , the duration is calculating from the outbound initiate time.
>>
>> How to get the call answer time?
>>
> If you're trying to do billing in-line you might as well stop right now and
> read the CDRs page on the wiki. Trying to calculate this from inside the
> session is silly.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Thu Mar 18 08:46:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 18 Mar 2010 11:46:11 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
Message-ID: <201003181146.11642.sos@sokhapkin.dyndns.org>

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

On Thursday 18 March 2010, Aloysius Lloyd wrote:
> Thank you for the suggestion. Yes I went to the direction then Stop. How to
> handle Hangup actions for every call?
> 
> For Example : I do not want to calculate the duration for all the calls or
> execute actions for all the calls. I need calculate for particular calls
> only.I tested the following ,* it execute for every calls hangup*. How to
> control or How to execute in some conditions.
> 
> 
> cdr_csv.conf.xml
> 
> 
> <param name="default-template" value="lua"/>
> 
> <template name="lua">${lua(onhangup.lua)}</template>
> 
> On Thu, Mar 18, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:
> > On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd 
<lloyd.aloysius at gmail.com>wrote:
> >> Hi All,
> >>
> >> The following lua script for calculate the call duration. But the issue
> >> here , the duration is calculating from the outbound initiate time.
> >>
> >> How to get the call answer time?
> >
> > If you're trying to do billing in-line you might as well stop right now
> > and read the CDRs page on the wiki. Trying to calculate this from inside
> > the session is silly.
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From msc at freeswitch.org  Thu Mar 18 08:49:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:49:29 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
Message-ID: <87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>

On Thu, Mar 18, 2010 at 8:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thank you for the suggestion. Yes I went to the direction then Stop. How to
> handle Hangup actions for every call?
>
> For Example : I do not want to calculate the duration for all the calls or
> execute actions for all the calls. I need calculate for particular calls
> only.I tested the following ,* it execute for every calls hangup*. How to
> control or How to execute in some conditions.
>

Is there a problem with running it on every call? The overhead of a Lua
script is minimal compared to the actual call. I'm just curious what you
gain from selectively running the script.
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 09:05:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 12:05:00 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
Message-ID: <8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>

 for example in a 10000 calls only want to run 100 times. Do you think it is
a best practice run the Lua script for 9900 time for no reason.

Lloyd

On Thu, Mar 18, 2010 at 11:49 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 8:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Thank you for the suggestion. Yes I went to the direction then Stop. How
>> to handle Hangup actions for every call?
>>
>> For Example : I do not want to calculate the duration for all the calls or
>> execute actions for all the calls. I need calculate for particular calls
>> only.I tested the following ,* it execute for every calls hangup*. How to
>> control or How to execute in some conditions.
>>
>
> Is there a problem with running it on every call? The overhead of a Lua
> script is minimal compared to the actual call. I'm just curious what you
> gain from selectively running the script.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 18 08:57:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:57:37 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
	<eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
Message-ID: <87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>

On Thu, Mar 18, 2010 at 12:39 AM, Vladimir Elizarov <xengelpublicx at gmail.com
> wrote:

> If she leaves the queue, the queue will not be agents. A user simply
> intercepting the call should be able to transfer the call to your
> phone when the Secretary is not there.
>
I'm sorry but I don't understand. If the receptionist logs out of the queue
then calls that are queued up will go to other agents (if available) or stay
in queue until the receptionist gets back. Also, you can do the "agent
access" extension I mentioned earlier so that non-agents can take callers
out of the queue.
-MC
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From msc at freeswitch.org  Thu Mar 18 09:11:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 09:11:52 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
Message-ID: <87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>

On Thu, Mar 18, 2010 at 9:05 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

>  for example in a 10000 calls only want to run 100 times. Do you think it
> is a best practice run the Lua script for 9900 time for no reason.

Do you need to skip processing the CDR altogether for those other 9900
calls?
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 09:22:23 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 12:22:23 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
	<87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
Message-ID: <8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>

No.Need CDR for all calls.

I am thinking in the following way. Is this possible?

Application A - Need CDR values and cdr_csv.conf.xml use *onhangupA.lua*

Application B - Need CDR values and cdr_csv.conf.xml use *onhangupB.lua*

Application C - Need CDR values only.


Thanks
Lloyd

On Thu, Mar 18, 2010 at 12:11 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 9:05 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>  for example in a 10000 calls only want to run 100 times. Do you think it
>> is a best practice run the Lua script for 9900 time for no reason.
>
> Do you need to skip processing the CDR altogether for those other 9900
> calls?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at redbonez.net  Thu Mar 18 09:52:13 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 10:52:13 -0600
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
Message-ID: <021d01cac6bb$5d73e1e0$185ba5a0$@net>

If you are a Python fan you should check out Twisted and
http://code.google.com/p/eventsocket/

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, March 18, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

 

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
wrote:

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.  We
> are building a website that needs to interact with FS.  Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.  Does
> this all need to be done using scripting languages like Lua or Perl, or
can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.

 


I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC

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From phunk0000 at hotmail.com  Thu Mar 18 10:37:23 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 13:37:23 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <021d01cac6bb$5d73e1e0$185ba5a0$@net>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
Message-ID: <BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>

Thanks folks!  I believe we will be doing most of our coding in php.  We are
also going to have independent DB servers.. I was also wondering if putting
the FS database on a DB server is a god idea, or if it should be left on the
FS box.  'Preciate it yall.  Learning is phun

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, March 18, 2010 12:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

If you are a Python fan you should check out Twisted and
http://code.google.com/p/eventsocket/

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, March 18, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

 

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
wrote:

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.  We
> are building a website that needs to interact with FS.  Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.  Does
> this all need to be done using scripting languages like Lua or Perl, or
can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.

 


I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.791 / Virus Database: 271.1.1/2752 - Release Date: 03/18/10
03:33:00

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From brian at freeswitch.org  Thu Mar 18 10:41:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 12:41:59 -0500
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
Message-ID: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>

Not sure how much god has to do with this... Unless you're praying your DB doesn't crash.   Kidding aside.  Its really up to you it works both ways.  But I would recommend a dedicated network for interconnection and management and not run that over the public/sip facing interfaces.

/b

On Mar 18, 2010, at 12:37 PM, Todd wrote:

> Thanks folks!  I believe we will be doing most of our coding in php.  We are also going to have independent DB servers?. I was also wondering if putting the FS database on a DB server is a god idea, or if it should be left on the FS box.  ?Preciate it yall.  Learning is phun

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From msc at freeswitch.org  Thu Mar 18 10:43:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 10:43:13 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
	<87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
	<8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>
Message-ID: <87f2f3b91003181043h63735e06p80719bdaf816a71a@mail.gmail.com>

On Thu, Mar 18, 2010 at 9:22 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> No.Need CDR for all calls.
>
> I am thinking in the following way. Is this possible?
>
> Application A - Need CDR values and cdr_csv.conf.xml use *onhangupA.lua*
>
> Application B - Need CDR values and cdr_csv.conf.xml use *onhangupB.lua*
>
> Application C - Need CDR values only.
>

What you really need to be using is mod_xml_cdr and posting your CDRs to a
separate server and do all this stuff outside of the FS server. Using
mod_xml_cdr will definitely scale more easily than trying to play around
with crazy logic in the dialplan and CSV CDRs.

http://wiki.freeswitch.org/wiki/Mod_xml_cdr

It's a paradigm shift from CSV CDRs but when you realize just how powerful
it is you will never be satisfied with plain CSV CDRs any more.
-MC
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From jonas.gauffin at gmail.com  Thu Mar 18 10:46:25 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Mar 2010 18:46:25 +0100
Subject: [Freeswitch-users] Calls
Message-ID: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>

Hello,

Is it possible to list calls that are terminated in IVR's, are parked
or in early state?
"show calls" api just lists established calls, correct?

Best regards,
  Jonas


From phunk0000 at hotmail.com  Thu Mar 18 10:51:28 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 13:51:28 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>	<021d01cac6bb$5d73e1e0$185ba5a0$@net>	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
Message-ID: <BLU110-DS912F0C62DC672E60B3D59A72B0@phx.gbl>

Ha! GOOD, thanks, all the servers are nice and local with gigabit
cards/switches.  Thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 18, 2010 1:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

Not sure how much god has to do with this... Unless you're praying your DB
doesn't crash.   Kidding aside.  Its really up to you it works both ways.
But I would recommend a dedicated network for interconnection and management
and not run that over the public/sip facing interfaces.

 

/b

 

On Mar 18, 2010, at 12:37 PM, Todd wrote:





Thanks folks!  I believe we will be doing most of our coding in php.  We are
also going to have independent DB servers.. I was also wondering if putting
the FS database on a DB server is a god idea, or if it should be left on the
FS box.  'Preciate it yall.  Learning is phun

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.791 / Virus Database: 271.1.1/2752 - Release Date: 03/18/10
03:33:00

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From infos at madovsky.org  Thu Mar 18 10:59:51 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 13:59:51 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <74540B793B4E4E1B869CBD7D954C1383@MOBILEE1705>

Hi Anthony,

after 5 days of tests to resolve the hell of sendmail segmentation fault,
my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

    <param name="mailer-app" value="/usr/sbin/ssmtp"/>
    <param name="mailer-app-args" value="-t"/>

and apply the patch attached (no choice since it needs filename as argument and not STDIN).

hope it will help people who have sendmail segmentation fault problem.

Regards

Franck

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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




            -- 
            Anthony Minessale II

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

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

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



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


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





      -- 
      Anthony Minessale II

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

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

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



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


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  -- 
  Anthony Minessale II

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

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

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



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From msc at freeswitch.org  Thu Mar 18 11:03:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 11:03:30 -0700
Subject: [Freeswitch-users] Calls
In-Reply-To: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>
References: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>
Message-ID: <87f2f3b91003181103j60a7f91eu5826b38f5b2cb400@mail.gmail.com>

show channels

On Thu, Mar 18, 2010 at 10:46 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> Is it possible to list calls that are terminated in IVR's, are parked
> or in early state?
> "show calls" api just lists established calls, correct?
>
> Best regards,
>  Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steve.d.ward at gmail.com  Thu Mar 18 11:04:56 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Thu, 18 Mar 2010 14:04:56 -0400
Subject: [Freeswitch-users] mod_fifo recording with fifo_record_template
Message-ID: <4ea6e8f21003181104j3775155y4b33ef27798beac6@mail.gmail.com>

I'm looking to use fifo_record_template to specify uniquely named recording
files.  Here is the situation:

I'm using a fifo with on-hook members.  I would like each fifo call to be
recorded, with each recording-file named uniquely.

Thanks to some helpful IRC talk, I have learned that the following works
well:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/my_recording.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

However, this records each and every fifo call to the file (in my case):
/usr/local/freeswitch/recordings/my_recording.wav

Every new call overwrites the file.

Instead of a statically defined file name, I would like a unique file name
for every fifo call.

I don't know how to make reference to channel variables in this definition.
I tried the following, but it results in an error:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/{${uuid}.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

The error I see on the CLI when a call goes into the FIFO is this:

2010-03-18 13:51:11.289448 [CRIT] switch_channel.c:759 Invalid data
(${fifo_record_template} contains a variable)

It was also suggested to me that I escape the '$' character on the
variable.  Like this:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/{\${uuid}.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

This also resulted in:

2010-03-18 13:52:37.014742 [CRIT] switch_channel.c:759 Invalid data
(${fifo_record_template} contains a variable)

The recording file doesn't get generated.

Any recommendations on using fifo_record_template to generate recording file
names that are unique to each call?  Is there a way to get channel variables
to be evaluated and populate the value of fifo_record_template?

Or does fifo_record_template only support static values (meaning I'd have to
do a record_session in dial plan?)?

Thanks for any feedback.
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From infos at madovsky.org  Thu Mar 18 11:10:55 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 14:10:55 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
Message-ID: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>

Also no need to uninstall sendmail, just install ssmtp and that's it !
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 1:59 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Hi Anthony,

  after 5 days of tests to resolve the hell of sendmail segmentation fault,
  my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
  I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
  So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

  configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

      <param name="mailer-app" value="/usr/sbin/ssmtp"/>
      <param name="mailer-app-args" value="-t"/>

  and apply the patch attached (no choice since it needs filename as argument and not STDIN).

  hope it will help people who have sendmail segmentation fault problem.

  Regards

  Franck

    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 17, 2010 4:37 PM
    Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


    FYI if you made the change in your working copy you could generate a patch with

    svn diff > patch.txt

    After looking at your change, the problem is that many people depend on this working the way it currently is.
    Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



    On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

      Sorry Anthony I never made an official update in C language with diff patch
      so the change is at the line 620, I commented out the original line

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Wednesday, March 17, 2010 3:55 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        could you get a svn diff of the switch_utils.c so we can  see what the change was?




        On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

          Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
          it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

          I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

          Regards

          Franck
            ----- Original Message ----- 
            From: Madovsky 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 4:35 PM
            Subject: Re: [Freeswitch-users] voicemail options


            ok, I tried 50 times to leave a message,
            it worked only one time.
            It seems that sendmail needs to be free to send the email....

              ----- Original Message ----- 
              From: devel at thom.fr.eu.org 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 16, 2010 3:59 PM
              Subject: Re: [Freeswitch-users] voicemail options


              Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

              I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



              Fran?ois





              De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
              Envoy? : mardi 16 mars 2010 20:30
              ? : freeswitch-users at lists.freeswitch.org
              Objet : Re: [Freeswitch-users] voicemail options



              are you running FS as root or a regular user?
              maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



              On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

              I noticed that sendmail makes a segfault when FS is trying to send email

              and sendmail has some emails locked in folder queue....



              Regards



              Franck


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              -- 
              Anthony Minessale II

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

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

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



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


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              http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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        -- 
        Anthony Minessale II

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

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

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



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


        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    -- 
    Anthony Minessale II

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

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

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
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    pstn:+19193869900



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


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From will.traenkle at yahoo.com  Thu Mar 18 11:15:36 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Thu, 18 Mar 2010 11:15:36 -0700 (PDT)
Subject: [Freeswitch-users] dial by name directory // mod_directory?
Message-ID: <487344.64278.qm@web57604.mail.re1.yahoo.com>

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?

I heard this was a feature that was implemented or going to be implemented in freeswitch.

If it has been implemented, would you send me info on how to configure it?

If it has not been implemented, would you state when it is planned to be implemented?

I believe the module name is mod_directory, is that correct?

Thanks,

-Will



      
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From xengelpublicx at gmail.com  Thu Mar 18 11:17:17 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 21:17:17 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
	<eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
	<87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>
Message-ID: <eb0385731003181117o3140ec6bv325d14ccdd14654c@mail.gmail.com>

ok. I will try to realize, what I want with "agent access" extension.
Thanks for your help.

On Thu, Mar 18, 2010 at 6:57 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Mar 18, 2010 at 12:39 AM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>>
>> If she leaves the queue, the queue will not be agents. A user simply
>> intercepting the call should be able to transfer the call to your
>> phone when the Secretary is not there.
>
> I'm sorry but I don't understand. If the receptionist logs out of the queue
> then calls that are queued up will go to other agents (if available) or stay
> in queue until the receptionist gets back. Also, you can do the "agent
> access" extension I mentioned earlier so that non-agents can take callers
> out of the queue.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov


From msc at freeswitch.org  Thu Mar 18 11:28:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 11:28:44 -0700
Subject: [Freeswitch-users] dial by name directory // mod_directory?
In-Reply-To: <487344.64278.qm@web57604.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>

On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle
<will.traenkle at yahoo.com>wrote:

> Has freeswitch implemented the "Dial by Name" feature where a caller can
> enter the first x number of characters of an extensions first or last name
> and have the operator either connect the user to the result and/or state the
> result's extension number?
>
> I heard this was a feature that was implemented or going to be implemented
> in freeswitch.
>
> If it has been implemented, would you send me info on how to configure it?
>
> If it has not been implemented, would you state when it is planned to be
> implemented?
>
> I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report
back.
-MC
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From lists at redbonez.net  Thu Mar 18 11:41:19 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 12:41:19 -0600
Subject: [Freeswitch-users] Disable call waiting
Message-ID: <026101cac6ca$9b010830$d1031890$@net>

I apologize if this has been answered before, but I can't seem to find
anything on google. 

 

Is there a way to disable call waiting in FreeSWITCH? 

 

-Adam

 

 

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From brian at freeswitch.org  Thu Mar 18 11:46:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 13:46:06 -0500
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <026101cac6ca$9b010830$d1031890$@net>
References: <026101cac6ca$9b010830$d1031890$@net>
Message-ID: <DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>

FreeSWITCH is NOT involved in call waiting.  Thats an endpoint specific feature.

If you have an endpoint that doesn't let you see mod_limit.

/b

On Mar 18, 2010, at 1:41 PM, Adam Ford wrote:

> I apologize if this has been answered before, but I can?t seem to find anything on google.
>  
> Is there a way to disable call waiting in FreeSWITCH?
>  
> -Adam

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From lists at redbonez.net  Thu Mar 18 11:51:36 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 12:51:36 -0600
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>
References: <026101cac6ca$9b010830$d1031890$@net>
	<DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>
Message-ID: <027201cac6cc$0aa4c900$1fee5b00$@net>

That would explain the lack of info.  Thanks Brian

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 18, 2010 12:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Disable call waiting

 

FreeSWITCH is NOT involved in call waiting.  Thats an endpoint specific
feature.

 

If you have an endpoint that doesn't let you see mod_limit.

 

/b

 

On Mar 18, 2010, at 1:41 PM, Adam Ford wrote:





I apologize if this has been answered before, but I can't seem to find
anything on google.

 

Is there a way to disable call waiting in FreeSWITCH?

 

-Adam

 

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From anthony.minessale at gmail.com  Thu Mar 18 11:56:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 13:56:10 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268900370911-4755257.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
Message-ID: <191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>

didn't I ask you to type show channels between the calls once they were
established?
are these bypass media calls?


On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> You're right, sorry.
>
> I updated to trunk (rev 17031) and the problem is still here.
>
> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
> They
> contain comments and commands that I used.
>
> Scenario:
> 1st call is made and is parked. Then 2nd and 3rd calls are:
> 1) dialed
> 2) parked
> 3) bridged with 1st call
> 4) hanged up
>
> It is OK for now.
> Then a 4th call is dialed, parked, bridged and killed. After killing the
> 4th
> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>
>
> I hope this description and logs help resolve the issue.
> Please let me know if you need any more information.
>
> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From freeswitch-users-list at metik.com  Thu Mar 18 12:01:21 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Thu, 18 Mar 2010 15:01:21 -0400
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <026101cac6ca$9b010830$d1031890$@net>
References: <026101cac6ca$9b010830$d1031890$@net>
Message-ID: <4BA27881.3090205@metik.com>

I recommend addressing this issue via endpoint configuration. If you 
really must to do it, I have successfully used "Mod Limit 
(http://wiki.freeswitch.org/wiki/Mod_limit) to do it in an effort to 
band-aid a call waiting bug exhibited by a SIP-based DSLAM (until the 
manufacturer could fix it). Not desirable--but works.

-Metik


Adam Ford wrote:
>
> I apologize if this has been answered before, but I can?t seem to find 
> anything on google.
>
> Is there a way to disable call waiting in FreeSWITCH?
>
> -Adam
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Thu Mar 18 12:03:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:03:54 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
Message-ID: <191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>

even better

also do

sofia profile internal siptrace on

to include the sip traffic


On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> didn't I ask you to type show channels between the calls once they were
> established?
> are these bypass media calls?
>
>
>
> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
>
>>
>> You're right, sorry.
>>
>> I updated to trunk (rev 17031) and the problem is still here.
>>
>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>> They
>> contain comments and commands that I used.
>>
>> Scenario:
>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>> 1) dialed
>> 2) parked
>> 3) bridged with 1st call
>> 4) hanged up
>>
>> It is OK for now.
>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>> 4th
>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>
>>
>> I hope this description and logs help resolve the issue.
>> Please let me know if you need any more information.
>>
>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>> --
>> View this message in context:
>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Mar 18 12:08:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:08:19 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
Message-ID: <191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>

Never mind,

I was right the first time 6 emails ago:  It's plenty obvious in this trace:

You bridged the 2 legs and you hungup on the A-leg when B-leg was not
answered yet.

here is your dialstring that tells us the problem.


bgapi originate {originate_timeout=20}[ignore_early_media=true,max_forwards=
999999,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
sofia/gateway/prospect_gateway/13007310757871097512282 &park()


ignore_early_media=true is not valid inside [] brackets only the global
section {} brackets.

if you move it there then you will never have any unanswered channels to
work with.



On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> even better
>
> also do
>
> sofia profile internal siptrace on
>
> to include the sip traffic
>
>
>
> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> didn't I ask you to type show channels between the calls once they were
>> established?
>> are these bypass media calls?
>>
>>
>>
>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>> maciej.aniserowicz at gmail.com> wrote:
>>
>>>
>>> You're right, sorry.
>>>
>>> I updated to trunk (rev 17031) and the problem is still here.
>>>
>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>> They
>>> contain comments and commands that I used.
>>>
>>> Scenario:
>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>> 1) dialed
>>> 2) parked
>>> 3) bridged with 1st call
>>> 4) hanged up
>>>
>>> It is OK for now.
>>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>>> 4th
>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>>
>>>
>>> I hope this description and logs help resolve the issue.
>>> Please let me know if you need any more information.
>>>
>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>>> --
>>> View this message in context:
>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Mar 18 12:16:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:16:49 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
Message-ID: <191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>

furthermore,

 to make sure this is over,

I added

uuid_bridge_continue_on_cancel=true chanvar you can set on A-leg to cause it

to continue in the dp even in your incorrect scenario


On Thu, Mar 18, 2010 at 2:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Never mind,
>
> I was right the first time 6 emails ago:  It's plenty obvious in this
> trace:
>
> You bridged the 2 legs and you hungup on the A-leg when B-leg was not
> answered yet.
>
> here is your dialstring that tells us the problem.
>
>
> bgapi originate {originate_timeout=20}[
> ignore_early_media=true,max_forwards=999999
> ,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
> sofia/gateway/prospect_gateway/13007310757871097512282 &park()
>
>
> ignore_early_media=true is not valid inside [] brackets only the global
> section {} brackets.
>
> if you move it there then you will never have any unanswered channels to
> work with.
>
>
>
> On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> even better
>>
>> also do
>>
>> sofia profile internal siptrace on
>>
>> to include the sip traffic
>>
>>
>>
>> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> didn't I ask you to type show channels between the calls once they were
>>> established?
>>> are these bypass media calls?
>>>
>>>
>>>
>>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>>> maciej.aniserowicz at gmail.com> wrote:
>>>
>>>>
>>>> You're right, sorry.
>>>>
>>>> I updated to trunk (rev 17031) and the problem is still here.
>>>>
>>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>>> They
>>>> contain comments and commands that I used.
>>>>
>>>> Scenario:
>>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>>> 1) dialed
>>>> 2) parked
>>>> 3) bridged with 1st call
>>>> 4) hanged up
>>>>
>>>> It is OK for now.
>>>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>>>> 4th
>>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>>>
>>>>
>>>> I hope this description and logs help resolve the issue.
>>>> Please let me know if you need any more information.
>>>>
>>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>>>> --
>>>> View this message in context:
>>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From tjardick at vanderkraan.net  Thu Mar 18 13:02:31 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 21:02:31 +0100
Subject: [Freeswitch-users] moh background in Lua
Message-ID: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>

Hi.

I'm trying to create a lua script but would like a certain task (music  
on hold) to be 'backgrounded':

-- Loop while the call is active
while (session:ready() == true) do
    -- do something here
    session:execute("playback", moh)
    freeswitch.consoleLog("INFO","Started music on hold.\n");
    -- do some stuff while playing music
    session:sleep(30000);
   -- Maybe we need a way to stop music or playback will stop  
automatically?
   session:execute("playback", "/sounds/we_will_answer_asap.wav")
   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
end

"ofcourse" currently the only thin that happens is the music and my  
loop does not continue.
Is there a way to start/stop music in the background?

I guess in Asterisk language i will need the command StartMusicOnHold...

Thanks for any suggestions.

Regards,

Tj


From anthony.minessale at gmail.com  Thu Mar 18 13:10:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 15:10:31 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
Message-ID: <191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>

The dilemma is we also need solution that is compat with windows, bsd, mac
etc.

That is sort of interesting that the pipe is breaking, it would be nice to
figure out why since not everybody
seems to have it.

I am still wondering if it's a stack size running out issue in the child
thread from system.
Are you running FS as root or as a non-privileged user, i'm wondering
because maybe if its a regular user, its not able to increase the stack size
up to 8megs

maybe you can push in strace into the command so it will show why it breaks.



On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

>  Also no need to uninstall sendmail, just install ssmtp and that's it !
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 18, 2010 1:59 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> Hi Anthony,
>
> after 5 days of tests to resolve the hell of sendmail segmentation fault,
> my conclusion is there is definitely a problem with the combination of cat,
> pipe and sendmail or other mail app.
> I made a php script that fits the actual switch_utils.c exec, but it's the
> same, it works one time on 10.
> So the robustest solution  is to replace sendmail by ssmtp (yum install
> ssmtp or apt-get ssmtp etc...)
>
> configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the
> switch.conf.xml like this
>
>     <param name="mailer-app" value="/usr/sbin/ssmtp"/>
>     <param name="mailer-app-args" value="-t"/>
>
> and apply the patch attached (no choice since it needs filename as argument
> and not STDIN).
>
>
> hope it will help people who have sendmail segmentation fault problem.
>
> Regards
>
> Franck
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 17, 2010 4:37 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> FYI if you made the change in your working copy you could generate a patch
> with
>
> svn diff > patch.txt
>
> After looking at your change, the problem is that many people depend on
> this working the way it currently is.
> Would you consider making a script that mimics the current behavior of a
> mailer app and read the file from stdin and deliver the subsequent file so
> it is backwards-compatible with the way everything already works?
>
>
> On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Sorry Anthony I never made an official update in C language with diff
>> patch
>> so the change is at the line 620, I commented out the original line
>>
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Wednesday, March 17, 2010 3:55 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>> could you get a svn diff of the switch_utils.c so we can  see what the
>> change was?
>>
>>
>>
>> On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  Finally I succeed to send voicemail with sendmail through a little php
>>> script (could be Perl also)
>>> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
>>> doesn't exist.
>>>
>>> I attached my php script and switch_utils.c. Not sure if the FS
>>> developers will accept my modification
>>>
>>> Regards
>>>
>>> Franck
>>>
>>> ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Tuesday, March 16, 2010 4:35 PM
>>> *Subject:* Re: [Freeswitch-users] voicemail options
>>>
>>> ok, I tried 50 times to leave a message,
>>> it worked only one time.
>>> It seems that sendmail needs to be free to send the email....
>>>
>>>
>>> ----- Original Message -----
>>> *From:* devel at thom.fr.eu.org
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Tuesday, March 16, 2010 3:59 PM
>>> *Subject:* Re: [Freeswitch-users] voicemail options
>>>
>>>  Are you running debian, I have the same problem on debian x64 (already
>>> mentioned here
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
>>> and cannot fix it.
>>>
>>> I don?t know if it?s a priviledge problem, as I got the same results
>>> running FS with or without ?u.
>>>
>>>
>>>
>>> Fran?ois
>>>
>>>
>>>
>>>
>>>
>>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
>>> Minessale
>>> *Envoy? :* mardi 16 mars 2010 20:30
>>> *? :* freeswitch-users at lists.freeswitch.org
>>> *Objet :* Re: [Freeswitch-users] voicemail options
>>>
>>>
>>>
>>> are you running FS as root or a regular user?
>>> maybe there is a permission or resource limitation problem preventing it
>>> from getting the larger stack space necessary?
>>>
>>>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> I noticed that sendmail makes a segfault when FS is trying to send email
>>>
>>> and sendmail has some emails locked in folder queue....
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 18 13:19:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 15:19:11 -0500
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
Message-ID: <191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>

The script is executing in the foreground and is being executed on your
channel.

You cannot tell the channel to do something else because it's already
running your script so
it can only run something else on top of it.

you have 2 choices:

1) use the socket app in async mode and external scripts with an ESL plugin.
2) instead of executing the script on the session launch a script in a sep
thread with the current uuid
    ad an argument then execute park and let the script control the channel
remotely

Even better reconsider what you want to do and avoid working against the
grain of how the existing architecture is flowing and you will have the best
results.



On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <
tjardick at vanderkraan.net> wrote:

> Hi.
>
> I'm trying to create a lua script but would like a certain task (music
> on hold) to be 'backgrounded':
>
> -- Loop while the call is active
> while (session:ready() == true) do
>    -- do something here
>    session:execute("playback", moh)
>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>    -- do some stuff while playing music
>    session:sleep(30000);
>   -- Maybe we need a way to stop music or playback will stop
> automatically?
>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
> end
>
> "ofcourse" currently the only thin that happens is the music and my
> loop does not continue.
> Is there a way to start/stop music in the background?
>
> I guess in Asterisk language i will need the command StartMusicOnHold...
>
> Thanks for any suggestions.
>
> Regards,
>
> Tj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Thu Mar 18 13:19:45 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 18 Mar 2010 22:19:45 +0200
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <4BA27881.3090205@metik.com>
References: <026101cac6ca$9b010830$d1031890$@net> <4BA27881.3090205@metik.com>
Message-ID: <d361ca7b1003181319j2202c25aj7fbf62ef504104c8@mail.gmail.com>

We do this by checking the core DB for the user's status. We do so since we
have a logic to decide whether to allow call waiting, send busy or send to
voicemail. The user's prefference is set in a database.

                        regards, __Yehavi:

2010/3/18 Metik <freeswitch-users-list at metik.com>

> I recommend addressing this issue via endpoint configuration. If you
> really must to do it, I have successfully used "Mod Limit
> (http://wiki.freeswitch.org/wiki/Mod_limit) to do it in an effort to
> band-aid a call waiting bug exhibited by a SIP-based DSLAM (until the
> manufacturer could fix it). Not desirable--but works.
>
> -Metik
>
>
> Adam Ford wrote:
> >
> > I apologize if this has been answered before, but I can?t seem to find
> > anything on google.
> >
> > Is there a way to disable call waiting in FreeSWITCH?
> >
> > -Adam
> >
> > ------------------------------------------------------------------------
>  >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From m.sobkow at marketelsystems.com  Thu Mar 18 13:31:00 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 14:31:00 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
Message-ID: <4BA28D84.6040109@marketelsystems.com>

I know eavesdrop is an application, not an API, so you have to use it 
differently.

I have an operator dialing in on extension 5000 being bridged to a 
client, and I'm getting those lovely little echoes that confirm the call 
is bridged (both operator and client softphones are sharing my one 
headset and mic.)

In the dialplan, I parse out the extension when someone dials in to 
665000 (last 4 digits are the extension to monitor), look up the 
appropriate operator UUID in my mnesia/Erlang tables, receive the "call" 
callback from Freeswitch, and build a proplists replacement with the 
execute-app-name set to "eavesdrop" and execute-app-arg set to the 
operator UUID.  I issue a sendmsg() command to Freeswitch, and it 
returns ok.

However, what happens to the extension that tried to eavesdrop is it 
gets two events: CHANNEL_PARK and SEND_MESSAGE.  The person making the 
eavesdrop call can't actually hear anything.

I've attached the Erlang callback module that the dialplan executes.  
The dialplan fragment is as follows:

<extension name=\"eavesdrop\">
    <condition field=\"destination_number\" 
expression=\"^66(\\d\\d\\d\\d)$\">
        <action application=\"answer\"/>
        <action application=\"set\" data=\"eavesdrop_extension=$1\" />
        <action application=\"erlang\" data=\"pbx_eavesdrop 
pursuit at testsrv\" />
    </condition>
</extension>

Unfortunately I can't think of any way to look up the UUID other than 
using the mnesia tables from the application.  When I tried the sample 
dialplans from Freeswitch, the eavesdrop module itself _did_ work.  I'm 
just having trouble cobbling together the "glue" for Erlang.

P.S. This is the last Freeswitch feature I have to code for this 
release, other than niggling cleanups like adding a few more voice 
prompts, which is easy enough as I've already sample code on how to do 
it.  With a little luck we'll make our April 1st ready-for-testing 
deadline!  We wouldn't have made it without Andrew's hints, tips, and 
tricks for using Erlang with Freeswitch. Thanks a lot! :D

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

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From fraserredmond at gmail.com  Thu Mar 18 13:45:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 18 Mar 2010 20:45:19 +0000
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl> 
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com> 
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com> 
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl> 
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
Message-ID: <c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>

Is there any advantage at all to putting the core database on your own
external db? Or is it just a control freak thing? I was looking at this the
other day, and only ever read that it was possible, not anything about why
you'd want to do it.

Cheers,
Fraser




On Thu, Mar 18, 2010 at 5:41 PM, Brian West <brian at freeswitch.org> wrote:

>  Its really up to you it works both ways.  But I would recommend a
> dedicated network for interconnection and management and not run that over
> the public/sip facing interfaces.
>
> /b
>
> On Mar 18, 2010, at 12:37 PM, Todd wrote:
>
> Thanks folks!  I believe we will be doing most of our coding in php.  We
> are also going to have independent DB servers?. I was also wondering if
> putting the FS database on a DB server is a god idea, or if it should be
> left on the FS box.  ?Preciate it yall.  Learning is phun
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tjardick at vanderkraan.net  Thu Mar 18 14:02:52 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 22:02:52 +0100
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
	<191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
Message-ID: <D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>

Hi Anthony,

Thanks for the quick and detailed reply.

Could i pick your brain on the last remark, what i'd like to do is:

- play a configurable time of hold music (30 seconds in the example)  
but this will be a variable later
- play a message (and optionally detect dtmf if needed)
- if no dtmf then continue music

and loop the above.

So basically creating an inbound holding point for people to wait  
before their call is answered, but give them the option to leave the  
'queue' to direct them else where.

Would in the async ESL i be able to stop the stream of music or would  
i need to do a bridge to local/9999 and unbridge to continue when i  
want to? Would ESL be the best way to
do this ?

Thanks for the help so far.

Regards,

Tj


On 18 Mar 2010, at 21:19, Anthony Minessale wrote:

> The script is executing in the foreground and is being executed on  
> your channel.
>
> You cannot tell the channel to do something else because it's  
> already running your script so
> it can only run something else on top of it.
>
> you have 2 choices:
>
> 1) use the socket app in async mode and external scripts with an ESL  
> plugin.
> 2) instead of executing the script on the session launch a script in  
> a sep thread with the current uuid
>     ad an argument then execute park and let the script control the  
> channel remotely
>
> Even better reconsider what you want to do and avoid working against  
> the grain of how the existing architecture is flowing and you will  
> have the best results.
>
>
>
> On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <tjardick at vanderkraan.net 
> > wrote:
> Hi.
>
> I'm trying to create a lua script but would like a certain task (music
> on hold) to be 'backgrounded':
>
> -- Loop while the call is active
> while (session:ready() == true) do
>    -- do something here
>    session:execute("playback", moh)
>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>    -- do some stuff while playing music
>    session:sleep(30000);
>   -- Maybe we need a way to stop music or playback will stop
> automatically?
>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
> end
>
> "ofcourse" currently the only thin that happens is the music and my
> loop does not continue.
> Is there a way to start/stop music in the background?
>
> I guess in Asterisk language i will need the command  
> StartMusicOnHold...
>
> Thanks for any suggestions.
>
> Regards,
>
> Tj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From andrew at hijacked.us  Thu Mar 18 14:11:39 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Mar 2010 17:11:39 -0400
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <4BA28D84.6040109@marketelsystems.com>
References: <4BA28D84.6040109@marketelsystems.com>
Message-ID: <20100318211138.GF28551@hijacked.us>

I do it like this:

freeswitch:bgapi(Fnode, originate, "user/agentname" ++ " &eavesdrop(" ++ UUID ++ ")")

But user/agentname can be any FS dialstring (SIP/... or whatever). It
dials the endpoint and then the b leg is the eavesdrop() call.

Andrew


From msc at freeswitch.org  Thu Mar 18 14:12:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 14:12:32 -0700
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>
Message-ID: <87f2f3b91003181412j2859edb1mcd4a1939a57c624b@mail.gmail.com>

Doc'd here:
http://wiki.freeswitch.org/wiki/Variable_uuid_bridge_continue_on_cancel
-MC

On Thu, Mar 18, 2010 at 12:16 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> furthermore,
>
>  to make sure this is over,
>
> I added
>
> uuid_bridge_continue_on_cancel=true chanvar you can set on A-leg to cause
> it
> to continue in the dp even in your incorrect scenario
>
>
> On Thu, Mar 18, 2010 at 2:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Never mind,
>>
>> I was right the first time 6 emails ago:  It's plenty obvious in this
>> trace:
>>
>> You bridged the 2 legs and you hungup on the A-leg when B-leg was not
>> answered yet.
>>
>> here is your dialstring that tells us the problem.
>>
>>
>> bgapi originate {originate_timeout=20}[
>> ignore_early_media=true,max_forwards=999999
>> ,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
>> sofia/gateway/prospect_gateway/13007310757871097512282 &park()
>>
>>
>> ignore_early_media=true is not valid inside [] brackets only the global
>> section {} brackets.
>>
>> if you move it there then you will never have any unanswered channels to
>> work with.
>>
>>
>>
>> On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> even better
>>>
>>> also do
>>>
>>> sofia profile internal siptrace on
>>>
>>> to include the sip traffic
>>>
>>>
>>>
>>> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> didn't I ask you to type show channels between the calls once they were
>>>> established?
>>>> are these bypass media calls?
>>>>
>>>>
>>>>
>>>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>>>> maciej.aniserowicz at gmail.com> wrote:
>>>>
>>>>>
>>>>> You're right, sorry.
>>>>>
>>>>> I updated to trunk (rev 17031) and the problem is still here.
>>>>>
>>>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>>>> They
>>>>> contain comments and commands that I used.
>>>>>
>>>>> Scenario:
>>>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>>>> 1) dialed
>>>>> 2) parked
>>>>> 3) bridged with 1st call
>>>>> 4) hanged up
>>>>>
>>>>> It is OK for now.
>>>>> Then a 4th call is dialed, parked, bridged and killed. After killing
>>>>> the 4th
>>>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL
>>>>> cause.
>>>>>
>>>>>
>>>>> I hope this description and logs help resolve the issue.
>>>>> Please let me know if you need any more information.
>>>>>
>>>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit
>>>>> host.
>>>>> --
>>>>> View this message in context:
>>>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Mar 18 14:20:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 16:20:40 -0500
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
	<191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
	<D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>
Message-ID: <191c3a031003181420n267d32f0oce0c3282c17e3385@mail.gmail.com>

You could use mod_fifo with the chimelist configured and avoid trying to
replicate this manually.

you can certainly do it from ESL because you have a socket connection to the
channel and all the commands you send will be in the background.


On Thu, Mar 18, 2010 at 4:02 PM, Tjardick van der Kraan <
tjardick at vanderkraan.net> wrote:

> Hi Anthony,
>
> Thanks for the quick and detailed reply.
>
> Could i pick your brain on the last remark, what i'd like to do is:
>
> - play a configurable time of hold music (30 seconds in the example) but
> this will be a variable later
> - play a message (and optionally detect dtmf if needed)
> - if no dtmf then continue music
>
> and loop the above.
>
> So basically creating an inbound holding point for people to wait before
> their call is answered, but give them the option to leave the 'queue' to
> direct them else where.
>
> Would in the async ESL i be able to stop the stream of music or would i
> need to do a bridge to local/9999 and unbridge to continue when i want to?
> Would ESL be the best way to
> do this ?
>
> Thanks for the help so far.
>
> Regards,
>
> Tj
>
>
> On 18 Mar 2010, at 21:19, Anthony Minessale wrote:
>
> The script is executing in the foreground and is being executed on your
> channel.
>
> You cannot tell the channel to do something else because it's already
> running your script so
> it can only run something else on top of it.
>
> you have 2 choices:
>
> 1) use the socket app in async mode and external scripts with an ESL
> plugin.
> 2) instead of executing the script on the session launch a script in a sep
> thread with the current uuid
>     ad an argument then execute park and let the script control the channel
> remotely
>
> Even better reconsider what you want to do and avoid working against the
> grain of how the existing architecture is flowing and you will have the best
> results.
>
>
>
> On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <
> tjardick at vanderkraan.net> wrote:
>
>> Hi.
>>
>> I'm trying to create a lua script but would like a certain task (music
>> on hold) to be 'backgrounded':
>>
>> -- Loop while the call is active
>> while (session:ready() == true) do
>>    -- do something here
>>    session:execute("playback", moh)
>>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>>    -- do some stuff while playing music
>>    session:sleep(30000);
>>   -- Maybe we need a way to stop music or playback will stop
>> automatically?
>>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
>> end
>>
>> "ofcourse" currently the only thin that happens is the music and my
>> loop does not continue.
>> Is there a way to start/stop music in the background?
>>
>> I guess in Asterisk language i will need the command StartMusicOnHold...
>>
>> Thanks for any suggestions.
>>
>> Regards,
>>
>> Tj
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Thu Mar 18 14:24:41 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 17:24:41 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
Message-ID: <3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>

Ok, I read that there are some guys who had the same problem here.
I run FS as root, but sendmail as chroot smmsp user. You're right maybe
it's a thread child problem. I saw there is a windows condition already in your code ?
and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
I will experiment with strace tonight or tomorrow
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  The dilemma is we also need solution that is compat with windows, bsd, mac etc.

  That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
  seems to have it.

  I am still wondering if it's a stack size running out issue in the child thread from system.
  Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

  maybe you can push in strace into the command so it will show why it breaks.




  On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

    Also no need to uninstall sendmail, just install ssmtp and that's it !
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, March 18, 2010 1:59 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      Hi Anthony,

      after 5 days of tests to resolve the hell of sendmail segmentation fault,
      my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
      I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
      So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

      configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

          <param name="mailer-app" value="/usr/sbin/ssmtp"/>
          <param name="mailer-app-args" value="-t"/>

      and apply the patch attached (no choice since it needs filename as argument and not STDIN).
       
    hope it will help people who have sendmail segmentation fault problem.

    Regards

    Franck

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 4:37 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      FYI if you made the change in your working copy you could generate a patch with

      svn diff > patch.txt

      After looking at your change, the problem is that many people depend on this working the way it currently is.
      Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



      On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

        Sorry Anthony I never made an official update in C language with diff patch
        so the change is at the line 620, I commented out the original line

          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Wednesday, March 17, 2010 3:55 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


          could you get a svn diff of the switch_utils.c so we can  see what the change was?




          On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

            Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
            it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

            I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

            Regards

            Franck
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 16, 2010 4:35 PM
              Subject: Re: [Freeswitch-users] voicemail options


              ok, I tried 50 times to leave a message,
              it worked only one time.
              It seems that sendmail needs to be free to send the email....

                ----- Original Message ----- 
                From: devel at thom.fr.eu.org 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 16, 2010 3:59 PM
                Subject: Re: [Freeswitch-users] voicemail options


                Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                Fran?ois





                De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                Envoy? : mardi 16 mars 2010 20:30
                ? : freeswitch-users at lists.freeswitch.org
                Objet : Re: [Freeswitch-users] voicemail options



                are you running FS as root or a regular user?
                maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                I noticed that sendmail makes a segfault when FS is trying to send email

                and sendmail has some emails locked in folder queue....



                Regards



                Franck


                _______________________________________________
                FreeSWITCH-users mailing list
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                FreeSWITCH http://www.freeswitch.org/
                ClueCon http://www.cluecon.com/
                Twitter: http://twitter.com/FreeSWITCH_wire

                AIM: anthm
                MSN:anthony_minessale at hotmail.com
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          -- 
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          FreeSWITCH http://www.freeswitch.org/
          ClueCon http://www.cluecon.com/
          Twitter: http://twitter.com/FreeSWITCH_wire

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

          FreeSWITCH Developer Conference
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          iax:guest at conference.freeswitch.org/888
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          pstn:+19193869900



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

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

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
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      pstn:+19193869900



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

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

  FreeSWITCH Developer Conference
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From m.sobkow at marketelsystems.com  Thu Mar 18 14:36:53 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 15:36:53 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <20100318211138.GF28551@hijacked.us>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
Message-ID: <4BA29CF5.1050101@marketelsystems.com>

Ok, so that new originate command will give me another UUID, and I'll 
have to bridge it to the original call's UUID, correct?  i.e. Someone 
dials the 66xxxx extension, it calls back to Erlang and parks.  Erlang 
issues the originate command, returns a UUID for the new call, and I do 
a uuid_bridge so the caller can hear the eavesdrop.

The only thing I'm a little puzzled by is the user/agentname string.  
The caller is already on the phone, so how can Freeswitch call them?

Alternatively maybe I could set the operator UUID into a variable and 
establish the call through a dialplan extension, but I think that ends 
up with the same problem of the caller being busy.

I do have one other option -- we have a softphone built into our 
front-end application, so I could have a button press/menu selection on 
that application issue a command to Erlang, which would mean it could 
call the softphone because it's not busy trying to dial a special 66xxxx 
extension.  That actually might be the best solution, because then there 
is no "secret" extension in the dialplan that unauthorized users might 
learn about.  I just need to have one of our other developers provide an 
API to retrieve the configured extension/registration information for 
the softphone so it can pass along the extension to Erlang for building 
a dial string.  The only "downside" to that approach is that it 
_requires_ our client application to work, but I think our user 
community will be ok with that because managers need that application 
for a whole host of features, not just the softphone.

Andrew Thompson wrote:
> I do it like this:
>
> freeswitch:bgapi(Fnode, originate, "user/agentname" ++ " &eavesdrop(" ++ UUID ++ ")")
>
> But user/agentname can be any FS dialstring (SIP/... or whatever). It
> dials the endpoint and then the b leg is the eavesdrop() call.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


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



From lloydie.t at googlemail.com  Thu Mar 18 14:45:02 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Thu, 18 Mar 2010 21:45:02 +0000
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
Message-ID: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>

I eventually managed to get mod_xml_curl working for me. Now I trying to be
a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I
get an error when a phone tries to register.

2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
203 at 192.168.101.102]
You must define a domain called '192.168.101.102' in your directory and add
a user with the id="203" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
(REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
192.168.91.28

I think I am nearly there,but I suspect there is something wrong with the
resulting 'user directory' xml.
---------------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="User Directory">
<domain name="default">
<groups>
  <group name="default">
  <users>
  <user id="203">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      <param name="vm-password" value="1234"/>
      <param name="vm-disk-quota" value="30"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
      <variable name="accountcode" value="203"/>
      <variable name="day_bar" value="3"/>
      <variable name="night_bar" value="3"/>
      <variable name="effective_caller_id_name" value="Extension 203"/>
      <variable name="effective_caller_id_number" value="203"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_number}"/>
    </variables>
  </user>
  </users>
 </group>
</groups>
</domain>
</section>
</document>
--------------------------------------------------------------------

I think the Company Directory XML is OK
--------------------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="Company Directory">
<domain name="default">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="default"/>
  </variables>
</domain>
<domain name="phisystems.dom">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="phisystems.dom"/>
  </variables>
</domain>
</section>
</document>
--------------------------------------------------------------

I get the following result when I 'sofia status'
                     Name          Type
Data     State
=================================================================================================
                 internal       profile   sip:mod_sofia at 192.168.101.102:5060
RUNNING (0)
           phisystems.dom         alias
internal     ALIASED
            internal-ipv6       profile           sip:mod_sofia@[::1]:5060
RUNNING (0)
                  default         alias
internal     ALIASED
                 external       profile   sip:mod_sofia at 192.168.101.102:5080
RUNNING (0)
=================================================================================================

Thanks in advance
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From andrew at hijacked.us  Thu Mar 18 14:48:41 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Mar 2010 17:48:41 -0400
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <4BA29CF5.1050101@marketelsystems.com>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
	<4BA29CF5.1050101@marketelsystems.com>
Message-ID: <20100318214841.GG28551@hijacked.us>

On Thu, Mar 18, 2010 at 03:36:53PM -0600, Mark Sobkow wrote:
> Ok, so that new originate command will give me another UUID, and I'll 
> have to bridge it to the original call's UUID, correct?  i.e. Someone 
> dials the 66xxxx extension, it calls back to Erlang and parks.  Erlang 
> issues the originate command, returns a UUID for the new call, and I do 
> a uuid_bridge so the caller can hear the eavesdrop.
> 
> The only thing I'm a little puzzled by is the user/agentname string.  
> The caller is already on the phone, so how can Freeswitch call them?
> 
> Alternatively maybe I could set the operator UUID into a variable and 
> establish the call through a dialplan extension, but I think that ends 
> up with the same problem of the caller being busy.
> 
> I do have one other option -- we have a softphone built into our 
> front-end application, so I could have a button press/menu selection on 
> that application issue a command to Erlang, which would mean it could 
> call the softphone because it's not busy trying to dial a special 66xxxx 
> extension.  That actually might be the best solution, because then there 
> is no "secret" extension in the dialplan that unauthorized users might 
> learn about.  I just need to have one of our other developers provide an 
> API to retrieve the configured extension/registration information for 
> the softphone so it can pass along the extension to Erlang for building 
> a dial string.  The only "downside" to that approach is that it 
> _requires_ our client application to work, but I think our user 
> community will be ok with that because managers need that application 
> for a whole host of features, not just the softphone.
>
Oh, I see what you wanted. Why not just bridge the call to an inline
dialplan that calls eavesdrop() then? That way you don't have to serve
up an extension or anything annoying.

something like

uuid_transfer CallerUUID 'eavesdrop:EavesdropUUID' inline

?

Andrew


From kenfulmer at icstechnologysolutions.com  Thu Mar 18 15:05:29 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 18 Mar 2010 17:05:29 -0500
Subject: [Freeswitch-users] PaeTec
Message-ID: <012001cac6e7$1f711490$5e533db0$@com>

Has anyone created a gateway to connect to PaeTec? 

 

Thanks,

 

Ken Fulmer

 

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From pavera at gmail.com  Wed Mar 17 22:07:05 2010
From: pavera at gmail.com (Tom Christensen)
Date: Wed, 17 Mar 2010 23:07:05 -0600
Subject: [Freeswitch-users] openzap no audio?
Message-ID: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>

I've got freeswitch installed with a digium te110p hooked up to a PRI.

calls through the system work fine SIP->SIP, calling and receiving calls
over the PRI works, (IE the signalling part, a call can be made and incoming
calls come in over the PRI and I can route them based on DID), however,
there is no audio to either party.  I've checked, and there is an RTP stream
running while the call is up, there is no reason it would be blocked (IE,
SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
few posts that talked about no audio for a few seconds after a call is
connected, or that calls would connect and then drop after a few seconds.
 This is different, I can leave the call up for many minutes, it doesn't
drop, and audio never starts coming through.

I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
the latest release of zaptel first, freeswitch wouldn't even load with that.

I'm going to try libpri now I think as I've exhausted all the other options
I can think of.

Any input/ideas would be greatly appreciated.

-Tom
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From pavera at gmail.com  Thu Mar 18 13:28:02 2010
From: pavera at gmail.com (Tom Christensen)
Date: Thu, 18 Mar 2010 14:28:02 -0600
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
Message-ID: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>

In my quest to replace an unstable and deadlock prone asterisk solution I
was able to get calls working over my PRI last night using freeswitch,
native openzap, and dahdi 2.2.1.  However, in order to make it work, I had
to disable all echo cancellation in dahdi.  If any of the 4 echo
cancellation algo's were specified in /etc/dahdi/system.conf then there was
no audio at all on the calls.  Signaling worked, and calls could be
made/received, but no audio in either direction.  As soon as I disabled EC,
audio worked (although, there was a good amount of echo).

Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very
poor voice quality (robotic sounding, distortions, dropping in and out).

This experience leads me to believe that possibly openzap isn't ready for
prime time?  Would I have been better served to include libpri in the
technology stack?

At this stage my "solution" looks like leaving asterisk where it is, and
using it as a media gateway, passing calls off to freeswitch via SIP or IAX
and using freeswitch as the app/registration/feature server.  I don't know
if that will necessarily be better, but I've had good luck with asterisk and
call quality tuning (the PRI here has always had pretty bad echo, but I've
been able to get asterisk EC working quite well on it), and it seems like
asterisk has most of its issues with stability in
transferring/parking/hold/voicemail/music on hold...

I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap using
a digium te110p.  The PRI is an NI2 t1 if that matters...

Any ideas or recommendations of things to try would be greatly appreciated.
I'd really like to avoid the dual server situation if at all possible.

Thanks,
-Tom
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From m.sobkow at marketelsystems.com  Thu Mar 18 15:23:01 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 16:23:01 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <20100318214841.GG28551@hijacked.us>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
	<4BA29CF5.1050101@marketelsystems.com>
	<20100318214841.GG28551@hijacked.us>
Message-ID: <4BA2A7C5.4030509@marketelsystems.com>

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From Russell.Mosemann at cune.org  Thu Mar 18 17:48:46 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Thu, 18 Mar 2010 19:48:46 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
References: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
Message-ID: <AD5B87549E604DD3BBD73065A1EF6317@cune.pri>

Tom Christensen wrote:

> If any of the 4 echo
> cancellation algo's were specified in /etc/dahdi/system.conf then there
> was no audio at all on the calls.

I'm not glad this happened to you, but I'm glad to hear about the problem. I have the same problem, and I thought it was something specific to my setup. I even compiled OSLEC separately with the same results. The call completes, but there is absolutely no audio. It is completely silent. I suppose that is a pretty effective echo canceller. :-) I did not notice any errors in the console. If EC is removed, the audio works.

> Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very
> poor voice quality (robotic sounding, distortions, dropping in and out).

I have not experienced that. There seems to be a brief audio dropout every couple of seconds during a call, like a couple of packets are being dropped every now and then, but I haven't dug into it, yet.

> This experience leads me to believe that possibly openzap isn't ready for
> prime time?  Would I have been better served to include libpri in the
> technology stack?

I'm sorry to report that I am using libpri, but that's interesting. I'm curious about the source of the problem and why EC would be affected. I don't see a connection.

> I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap
> using a digium te110p.  The PRI is an NI2 t1 if that matters...

Exact same setup here. Hmm.

> Any ideas or recommendations of things to try would be greatly
> appreciated. I'd really like to avoid the dual server situation if at all
> possible.

Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all that bad here, but it would be nice to use EC.

----
Russell Mosemann



From gkuri at ieee.org  Thu Mar 18 19:48:15 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Mar 2010 19:48:15 -0700
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
Message-ID: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>

Hey guys, is the Caller ID data (number and name) supposed to be
passed between the phones when using SLA/SCA ? If you put a call on
hold on phone 1, and pick up the call on phone 2, number/name do not
show up. Likewise, if you go back to the same call on phone 1, the
name and number are not present. Is this normal? Do I smell a bounty
;) ?

Cheers,
Gabe


From infos at madovsky.org  Thu Mar 18 20:07:14 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 23:07:14 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
Message-ID: <581ED17E563D47C5AD4E91CBF91DC727@MOBILEE1705>

but I think that the patch can also work with sendmail
as default config too... it needs a test with someone who uses
sendmail without seg fault...
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 5:24 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Ok, I read that there are some guys who had the same problem here.
  I run FS as root, but sendmail as chroot smmsp user. You're right maybe
  it's a thread child problem. I saw there is a windows condition already in your code ?
  and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
  I will experiment with strace tonight or tomorrow
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 18, 2010 4:10 PM
    Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


    The dilemma is we also need solution that is compat with windows, bsd, mac etc.

    That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
    seems to have it.

    I am still wondering if it's a stack size running out issue in the child thread from system.
    Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

    maybe you can push in strace into the command so it will show why it breaks.




    On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

      Also no need to uninstall sendmail, just install ssmtp and that's it !
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 18, 2010 1:59 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        Hi Anthony,

        after 5 days of tests to resolve the hell of sendmail segmentation fault,
        my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
        I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
        So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

        configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

            <param name="mailer-app" value="/usr/sbin/ssmtp"/>
            <param name="mailer-app-args" value="-t"/>

        and apply the patch attached (no choice since it needs filename as argument and not STDIN).
         
      hope it will help people who have sendmail segmentation fault problem.

      Regards

      Franck

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Wednesday, March 17, 2010 4:37 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        FYI if you made the change in your working copy you could generate a patch with

        svn diff > patch.txt

        After looking at your change, the problem is that many people depend on this working the way it currently is.
        Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



        On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

          Sorry Anthony I never made an official update in C language with diff patch
          so the change is at the line 620, I commented out the original line

            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Wednesday, March 17, 2010 3:55 PM
            Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


            could you get a svn diff of the switch_utils.c so we can  see what the change was?




            On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

              Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
              it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

              I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

              Regards

              Franck
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 16, 2010 4:35 PM
                Subject: Re: [Freeswitch-users] voicemail options


                ok, I tried 50 times to leave a message,
                it worked only one time.
                It seems that sendmail needs to be free to send the email....

                  ----- Original Message ----- 
                  From: devel at thom.fr.eu.org 
                  To: freeswitch-users at lists.freeswitch.org 
                  Sent: Tuesday, March 16, 2010 3:59 PM
                  Subject: Re: [Freeswitch-users] voicemail options


                  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                  I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                  Fran?ois





                  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                  Envoy? : mardi 16 mars 2010 20:30
                  ? : freeswitch-users at lists.freeswitch.org
                  Objet : Re: [Freeswitch-users] voicemail options



                  are you running FS as root or a regular user?
                  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                  I noticed that sendmail makes a segfault when FS is trying to send email

                  and sendmail has some emails locked in folder queue....



                  Regards



                  Franck


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                  -- 
                  Anthony Minessale II

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

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

                  FreeSWITCH Developer Conference
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                  iax:guest at conference.freeswitch.org/888
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            -- 
            Anthony Minessale II

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

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

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



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        -- 
        Anthony Minessale II

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

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

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



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


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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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    -- 
    Anthony Minessale II

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

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

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



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From brian at freeswitch.org  Thu Mar 18 20:23:44 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 22:23:44 -0500
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
In-Reply-To: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
References: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
Message-ID: <6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>

I would post a bounty...   ;)

/b

On Mar 18, 2010, at 9:48 PM, Gabriel Kuri wrote:

> Hey guys, is the Caller ID data (number and name) supposed to be
> passed between the phones when using SLA/SCA ? If you put a call on
> hold on phone 1, and pick up the call on phone 2, number/name do not
> show up. Likewise, if you go back to the same call on phone 1, the
> name and number are not present. Is this normal? Do I smell a bounty
> ;) ?



From gkuri at ieee.org  Thu Mar 18 21:10:13 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Mar 2010 21:10:13 -0700
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
In-Reply-To: <6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>
References: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
	<6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>
Message-ID: <8b1c9cda1003182110v411a7a7k5ccd316cf2f3e140@mail.gmail.com>

About how much, what kind of work is required for something like this?

Cheers,
Gabe

On Thu, Mar 18, 2010 at 8:23 PM, Brian West <brian at freeswitch.org> wrote:
> I would post a bounty... ? ;)
>
> /b
>
> On Mar 18, 2010, at 9:48 PM, Gabriel Kuri wrote:
>
>> Hey guys, is the Caller ID data (number and name) supposed to be
>> passed between the phones when using SLA/SCA ? If you put a call on
>> hold on phone 1, and pick up the call on phone 2, number/name do not
>> show up. Likewise, if you go back to the same call on phone 1, the
>> name and number are not present. Is this normal? Do I smell a bounty
>> ;) ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Thu Mar 18 21:32:29 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 19 Mar 2010 11:32:29 +0700
Subject: [Freeswitch-users] Indian and mod_say
Message-ID: <8ccbff061003182132x697f6539jf7b5d6934c0924a9@mail.gmail.com>

Can someone guide me about Indian language? i want to create mod_say_in

BG

Dome C.


From infos at madovsky.org  Thu Mar 18 21:55:40 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 00:55:40 -0400
Subject: [Freeswitch-users] variables in gateway profile
Message-ID: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>

Is it possible to create <variable> in gateway profiles like users profiles ?

Thanks

Franck
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From will.traenkle at yahoo.com  Fri Mar 19 00:42:08 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Mar 2010 00:42:08 -0700 (PDT)
Subject: [Freeswitch-users] dial by name directory // mod_directory?
In-Reply-To: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
Message-ID: <787610.94105.qm@web57616.mail.re1.yahoo.com>

Thanks Michael.

I will test drive it and report back. I will also be testing incoming fax and conference calls as well next week.

Cheers,

-Will




________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com> wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?
>
>I heard this was a feature that was implemented or going to be implemented in freeswitch.
>
>If it has been implemented, would you send me info on how to configure it?
>
>If it has not been implemented, would you state when it is planned to be implemented?
>
>I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report back.
-MC


      
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From lakindia89 at gmail.com  Fri Mar 19 01:27:07 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 19 Mar 2010 13:57:07 +0530
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
Message-ID: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>

Hi all,
I've updated to the latest SVN 17048.
Here is my dialplan:
<include>
        <context name="default">
                <extension name="outbound_soc">
                        <condition field="destination_number"
expression="^.*$">
                                <action application="ring_ready" />
                                <action application="set"
data="continue_on_fail=true"/>
                                <action application="set"
data="bypass_media=true"/>
                                <action application="set"
data="ignore_early_media=true"/>
                                <action application="set"
data="exec_after_bridge_app=park"/>
                                <action application="socket" data="
0.0.0.0:8447 async full"/>
                        </condition>
                </extension>
        </context>
</include>

>From a nc session I issued the following commands.
connect

sendmsg
call-command: execute
execute-app-name: answer

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: user/1010


I (user/1001) is able to hear the voice of the bridged party (user/1010).
But user/1010 is unable to hear my voice.

If I set the bypass_media=false, then both can hear the voice.

I wanted to know why when bypass_media=true, the user/1010 cannot hear my
voice.
Here is the freeswitch log:
http://pastebin.freeswitch.org/12484
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From jason at jasonjgw.net  Fri Mar 19 01:51:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Mar 2010 19:51:57 +1100
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
In-Reply-To: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
References: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
Message-ID: <20100319085157.GA18092@jdc.jasonjgw.net>

lakshmanan ganapathy <lakindia89 at gmail.com> wrote:
> I wanted to know why when bypass_media=true, the user/1010 cannot hear my
> voice.

It's probably due to NAT between the two end-points, or some other network
issue. Have you used Wireshark or a similar packet analysis tool to determine
whether RTP packets are being sent out by both end-points to the correct
addresses?

If they are, then you'll need to find out why the packets in one direction
fail to reach the destination.



From m.krivushin at imarto.net  Fri Mar 19 02:22:35 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Fri, 19 Mar 2010 15:22:35 +0600
Subject: [Freeswitch-users] Video pass problem
In-Reply-To: <FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>
References: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
	<FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>
Message-ID: <5be734a51003190222w35f4db78he61ba6a138b41943@mail.gmail.com>

We use two different OSes - stable debian and bleeding edge ubuntu, but
problem live in both.
I think problem not in bleeding edge OS, but in way of processing rtp by FS.
I'm not sure, but think problem in renumerating of rtp packets in video
processing in mod_h26x.

If we have video problems with lost packets, after FS we have not this info,
its mascarade losts by renumerating. And, I suppose, decoding part cant
understand video with masacarding of lost frames.

All work fine with proxy mode, now, after small patching of old hack in
switch_rtp.c, tnx to anthm. But my billing module dont work in proxy mode
now, and I hear that proxy mode is hack for t.38.

2010/3/5 Michael Jerris <mike at jerris.com>

> double check if the behavior on this is the same with svn trunk of
> FreeSWITCH on something other than a bleeding edge operating system.
>
> Mike
>
> On Feb 25, 2010, at 7:59 AM, Mikhail Krivushin wrote:
>
> Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and
> see that we have 1280 video packets input, and only 560 passed to B leg.
> Anyone can point me to right direction? I can send pcap file by request.
>
> Most time I have black screen, and then one frame can appear, and I see
> frozen picture pair minutes, and then other frozen picture. We have not
> network issues, we have good video, when bypass FS. We have not perfomance
> troubles to.
>
> We have ubuntu 9.10 x64, and powerfull server board. uname:
> Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009
> x86_64 GNU/Linux
>
> config:
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h
> @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h
> @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From devel at thom.fr.eu.org  Fri Mar 19 03:51:13 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Fri, 19 Mar 2010 11:51:13 +0100
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
Message-ID: <e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>



Actually, I do not seem to have the same problem. I don't get the broken
pipe error, nor do I get any error in FS log or console (the error message
in log disappeared after some upgrade but I don't remember which).


However, what I can tell is that I get the problem as root AND as non
root. I can also tell that sending the exact same message with the exact
same command line as my freeswitch user work (sudo -u freeswitch cat
savedmail | sendmail -f freeswitch at mydomain -t some.user at mydomain). 

This
does not seem to be an environment problem either (I could not seem
significant differences between the 2). 

I'm about to strace what happens
too. 

Fran?ois 

On Thu, 18 Mar 2010 17:24:41 -0400, "Madovsky"  wrote: 
Ok, I read that there are some guys who had the same problem here. I run FS
as root, but sendmail as chroot smmsp user. You're right maybe it's a
thread child problem. I saw there is a windows condition already in your
code ? and for mac it should work since ssmtp is in standard in all
package, even for solaris and hp-ux. I will experiment with strace tonight
or tomorrow  ----- Original Message ----- FROM: Anthony Minessale [1] TO:
freeswitch-users at lists.freeswitch.org [2] SENT: Thursday, March 18, 2010
4:10 PM SUBJECT: Re: [Freeswitch-users] voicemail options [temporarly
solved]  The dilemma is we also need solution that is compat with windows,
bsd, mac etc.

That is sort of interesting that the pipe is breaking, it
would be nice to figure out why since not everybody
seems to have it.

I am
still wondering if it's a stack size running out issue in the child thread
from system.
Are you running FS as root or as a non-privileged user, i'm
wondering because maybe if its a regular user, its not able to increase the
stack size up to 8megs 

maybe you can push in strace into the command so
it will show why it breaks.

 On Thu, Mar 18, 2010 at 1:10 PM, Madovsky 
wrote:
   Also no need to uninstall sendmail, just install ssmtp and that's
it !   ----- Original Message ----- FROM: Madovsky [4] TO:
freeswitch-users at lists.freeswitch.org [5]    SENT: Thursday, March 18, 2010
1:59 PM SUBJECT: Re: [Freeswitch-users] voicemail options [temporarly
solved]  Hi Anthony,     after 5 days of tests to resolve the hell of
sendmail segmentation fault,   my conclusion is there is definitely a
problem with the combination of cat, pipe and sendmail or other mail app.  
I made a php script that fits the actual switch_utils.c exec, but it's the
same, it works one time on 10.   So the robustest solution is to replace
sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)      
configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the
switch.conf.xml like this        

and apply the patch attached (no choice
since it needs filename as argument and not STDIN).              hope it
will help people who have sendmail segmentation fault problem.      
Regards       Franck         ----- Original Message -----    FROM: Anthony
Minessale [6]    TO: freeswitch-users at lists.freeswitch.org [7]    SENT:
Wednesday, March 17, 2010 4:37 PM   SUBJECT: Re: [Freeswitch-users]
voicemail options [temporarly solved]   
 FYI if you made the change in
your working copy you could generate a patch with

svn diff >
patch.txt

After looking at your change, the problem is that many people
depend on this working the way it currently is.
Would you consider making a
script that mimics the current behavior of a mailer app and read the file
from stdin and deliver the subsequent file so it is backwards-compatible
with the way everything already works?

 On Wed, Mar 17, 2010 at 3:26 PM,
Madovsky  wrote:
     Sorry Anthony I never made an official update in C
language with diff patch   so the change is at the line 620, I commented
out the original line           ----- Original Message -----    FROM:
Anthony Minessale [9]    TO: freeswitch-users at lists.freeswitch.org [10]    
     SENT: Wednesday, March 17, 2010 3:55 PM   SUBJECT: Re:
[Freeswitch-users] voicemail options [temporarly solved]   
 could you get
a svn diff of the switch_utils.c so we can see what the change was?

 On
Wed, Mar 17, 2010 at 2:46 PM, Madovsky  wrote:
     Finally I succeed to
send voicemail with sendmail through a little php script (could be Perl
also)   it seems that sendmail 8.14.3 has a bug. But call it from PHP the
bug doesn't exist.       I attached my php script and switch_utils.c. Not
sure if the FS developers will accept my modification       Regards      
Franck     ----- Original Message -----    FROM: Madovsky [12]    TO:
freeswitch-users at lists.freeswitch.org [13]    SENT: Tuesday, March 16, 2010
4:35 PM   SUBJECT: Re: [Freeswitch-users] voicemail options   
   ok, I
tried 50 times to leave a message,   it worked only one time.   It seems
that sendmail needs to be free to send the email....         ----- Original
Message -----    FROM: devel at thom.fr.eu.org [14]    TO:
freeswitch-users at lists.freeswitch.org [15]    SENT: Tuesday, March 16, 2010
3:59 PM   SUBJECT: Re: [Freeswitch-users] voicemail options   

Are you
running debian, I have the same problem on debian x64 (already mentioned
here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html
[16]) and cannot fix it.   

I don't know if it's a priviledge problem, as
I got the same results running FS with or without -u.   

Fran?ois   

DE :
freeswitch-users-bounces at lists.freeswitch.org [17]
[mailto:freeswitch-users-bounces at lists.freeswitch.org [18]] DE LA PART DE
Anthony Minessale
ENVOY : mardi 16 mars 2010 20:30
 :
freeswitch-users at lists.freeswitch.org [19]
OBJET : Re: [Freeswitch-users]
voicemail options    

are you running FS as root or a regular user?
maybe
there is a permission or resource limitation problem preventing it from
getting the larger stack space necessary?

On Tue, Mar 16, 2010 at 2:18 PM,
Madovsky  wrote:       

I noticed that sendmail makes a segfault when FS
is trying to send email      

and sendmail has some emails locked in
folder queue....      

Regards      

Franck    


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[2]
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[3]
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[27] http://twitter.com/FreeSWITCH_wire
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[44] http://twitter.com/FreeSWITCH_wire
[45]
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[60]
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[61] http://twitter.com/FreeSWITCH_wire
[62]
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[63]
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[65]
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[77]
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[78] http://twitter.com/FreeSWITCH_wire
[79]
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[80]
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[81]
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-------------- next part --------------
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From infos at madovsky.org  Fri Mar 19 04:41:30 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 07:41:30 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
Message-ID: <49B3842E181347948484F22AB2FABBD0@MOBILEE1705>

I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)
  ----- Original Message ----- 
  From: Fran?ois Legal 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 6:51 AM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Actually, I do not seem to have the same problem. I don't get the broken pipe error, nor do I get any error in FS log or console (the error message in log disappeared after some upgrade but I don't remember which).

  However, what I can tell is that I get the problem as root AND as non root. I can also tell that sending the exact same message with the exact same command line as my freeswitch user work (sudo -u freeswitch cat savedmail | sendmail -f freeswitch at mydomain -t some.user at mydomain).

  This does not seem to be an environment problem either (I could not seem significant differences between the 2).

  I'm about to strace what happens too.



  Fran?ois



  On Thu, 18 Mar 2010 17:24:41 -0400, "Madovsky" <infos at madovsky.org> wrote:

    Ok, I read that there are some guys who had the same problem here.
    I run FS as root, but sendmail as chroot smmsp user. You're right maybe
    it's a thread child problem. I saw there is a windows condition already in your code ?
    and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
    I will experiment with strace tonight or tomorrow
      ----- Original Message -----
      From: Anthony Minessale
      To: freeswitch-users at lists.freeswitch.org
      Sent: Thursday, March 18, 2010 4:10 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]
      The dilemma is we also need solution that is compat with windows, bsd, mac etc.

      That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
      seems to have it.

      I am still wondering if it's a stack size running out issue in the child thread from system.
      Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

      maybe you can push in strace into the command so it will show why it breaks.




      On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

        Also no need to uninstall sendmail, just install ssmtp and that's it !
          ----- Original Message -----
          From: Madovsky
          To: freeswitch-users at lists.freeswitch.org
          Sent: Thursday, March 18, 2010 1:59 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]
          Hi Anthony,

          after 5 days of tests to resolve the hell of sendmail segmentation fault,
          my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
          I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
          So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

          configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

              
              

          and apply the patch attached (no choice since it needs filename as argument and not STDIN).
           
        hope it will help people who have sendmail segmentation fault problem.

        Regards

        Franck

          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Wednesday, March 17, 2010 4:37 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


          FYI if you made the change in your working copy you could generate a patch with

          svn diff > patch.txt

          After looking at your change, the problem is that many people depend on this working the way it currently is.
          Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



          On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

            Sorry Anthony I never made an official update in C language with diff patch
            so the change is at the line 620, I commented out the original line

              ----- Original Message ----- 
              From: Anthony Minessale 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Wednesday, March 17, 2010 3:55 PM
              Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


              could you get a svn diff of the switch_utils.c so we can  see what the change was?




              On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

                Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
                it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

                I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

                Regards

                Franck
                  ----- Original Message ----- 
                  From: Madovsky 
                  To: freeswitch-users at lists.freeswitch.org 
                  Sent: Tuesday, March 16, 2010 4:35 PM
                  Subject: Re: [Freeswitch-users] voicemail options


                  ok, I tried 50 times to leave a message,
                  it worked only one time.
                  It seems that sendmail needs to be free to send the email....

                    ----- Original Message ----- 
                    From: devel at thom.fr.eu.org 
                    To: freeswitch-users at lists.freeswitch.org 
                    Sent: Tuesday, March 16, 2010 3:59 PM
                    Subject: Re: [Freeswitch-users] voicemail options


                    Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                    I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                    Fran?ois





                    De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                    Envoy? : mardi 16 mars 2010 20:30
                    ? : freeswitch-users at lists.freeswitch.org
                    Objet : Re: [Freeswitch-users] voicemail options




                    are you running FS as root or a regular user?
                    maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                    On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                    I noticed that sendmail makes a segfault when FS is trying to send email

                    and sendmail has some emails locked in folder queue....




                    Regards




                    Franck


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


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





      -- 
      Anthony Minessale II

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

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

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





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




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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From rentmycoder at gmail.com  Fri Mar 19 06:01:11 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Fri, 19 Mar 2010 14:01:11 +0100
Subject: [Freeswitch-users] how to export variables from b-leg to a-leg?
Message-ID: <50e456911003190601i395d108cg55c98a214e84eeae@mail.gmail.com>

Hi guys,
Freeswitch Rocks!

Does anybody know how to export variables from the b-leg to a-leg...
I start a bridge from lua on an outgoing gateway and on the answer of
the b-leg I would like to set a variable on the a-leg...

<action application="export" data="nolocal:execute_on_answer=lua
cc.lua extensions_answer ${uuid}"/>
<bridge ....>

from cc.lua I try:
session:execute("export", "foo=123");

but no success...
any hint?


From brian at freeswitch.org  Fri Mar 19 06:20:59 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 08:20:59 -0500
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
In-Reply-To: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
References: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
Message-ID: <D5BECA6B-086E-45AC-940A-D48DB39C35B2@freeswitch.org>

Because you can't use Proxy Media in this case.  You do realize that FreeSWITCH by default will proxy the media thru the switch.  Proxy media is a different ball of wax and has exact ONE USE CASE.  Anyone using Proxy media and isn't doing T.38/UDPTL needs to stop using it.  Its only for T.38 passthru so that FreeSWITCH doesn't look at the contents of the packet and just passes them along.

1. Default Media Passes thru FreeSWITCH and parses the RTP packets up into the switch.

2. Proxy Media passes thru FreeSWITCH but FreeSWITCH can't touch the media because it could be formats we don't know about ie UDPTL

3. Bypass Media keeps the signaling on FreeSWITCH but makes the media go P2P.


The bottom line is YOU SHOULDN'T BE USING PROXY MEDIA.

/b

On Mar 19, 2010, at 3:27 AM, lakshmanan ganapathy wrote:

> Hi all,
> I've updated to the latest SVN 17048.
> Here is my dialplan:
> <include>
>         <context name="default">
>                 <extension name="outbound_soc">
>                         <condition field="destination_number" expression="^.*$">
>                                 <action application="ring_ready" />
>                                 <action application="set" data="continue_on_fail=true"/>
>                                 <action application="set" data="bypass_media=true"/>
>                                 <action application="set" data="ignore_early_media=true"/>
>                                 <action application="set" data="exec_after_bridge_app=park"/>
>                                 <action application="socket" data="0.0.0.0:8447 async full"/>
>                         </condition>
>                 </extension>
>         </context>
> </include>
> 
> From a nc session I issued the following commands.
> connect
> 
> sendmsg
> call-command: execute
> execute-app-name: answer
> 
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: user/1010
> 
> 
> I (user/1001) is able to hear the voice of the bridged party (user/1010). But user/1010 is unable to hear my voice.
> 
> If I set the bypass_media=false, then both can hear the voice.
> 
> I wanted to know why when bypass_media=true, the user/1010 cannot hear my voice.
> Here is the freeswitch log:
> http://pastebin.freeswitch.org/12484
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From testeador01 at gmail.com  Fri Mar 19 06:39:37 2010
From: testeador01 at gmail.com (Milena)
Date: Fri, 19 Mar 2010 08:39:37 -0500
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
In-Reply-To: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
References: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
Message-ID: <c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>

Hello,

None of them are ok for the domain you want to register to, when you set up
multiple domains, you have to return the results with the domain name you're
giving information about, make it return another xml with
<domain name="192.168.101.102">

2010/3/18 lloyd thomas <lloydie.t at googlemail.com>

> I eventually managed to get mod_xml_curl working for me. Now I trying to be
> a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
> wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I
> get an error when a phone tries to register.
>
> 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
> 203 at 192.168.101.102]
> You must define a domain called '192.168.101.102' in your directory and add
> a user with the id="203" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
> 192.168.91.28
>
> I think I am nearly there,but I suspect there is something wrong with the
> resulting 'user directory' xml.
> ---------------------------------------------------------------
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="User Directory">
> <domain name="default">
> <groups>
>   <group name="default">
>   <users>
>   <user id="203">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>       <param name="vm-password" value="1234"/>
>       <param name="vm-disk-quota" value="30"/>
>     </params>
>     <variables>
>       <variable name="user_context" value="default"/>
>       <variable name="accountcode" value="203"/>
>       <variable name="day_bar" value="3"/>
>       <variable name="night_bar" value="3"/>
>       <variable name="effective_caller_id_name" value="Extension 203"/>
>       <variable name="effective_caller_id_number" value="203"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_number}"/>
>     </variables>
>   </user>
>   </users>
>  </group>
> </groups>
> </domain>
> </section>
> </document>
> --------------------------------------------------------------------
>
> I think the Company Directory XML is OK
> --------------------------------------------------------------------
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="Company Directory">
> <domain name="default">
>   <params>
>     <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>   </params>
>   <variables>
>     <variable name="record_stereo" value="true"/>
>     <variable name="default_gateway" value="$${default_provider}"/>
>     <variable name="default_areacode" value="$${default_areacode}"/>
>     <variable name="transfer_fallback_extension" value="operator"/>
>     <variable name="user_context" value="default"/>
>   </variables>
> </domain>
> <domain name="phisystems.dom">
>   <params>
>     <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>   </params>
>   <variables>
>     <variable name="record_stereo" value="true"/>
>     <variable name="default_gateway" value="$${default_provider}"/>
>     <variable name="default_areacode" value="$${default_areacode}"/>
>     <variable name="transfer_fallback_extension" value="operator"/>
>     <variable name="user_context" value="phisystems.dom"/>
>   </variables>
> </domain>
> </section>
> </document>
> --------------------------------------------------------------
>
> I get the following result when I 'sofia status'
>                      Name          Type
> Data     State
>
> =================================================================================================
>                  internal       profile
> sip:mod_sofia at 192.168.101.102:5060   RUNNING (0)
>            phisystems.dom         alias
> internal     ALIASED
>             internal-ipv6       profile           sip:mod_sofia@[::1]:5060
> RUNNING (0)
>                   default         alias
> internal     ALIASED
>                  external       profile
> sip:mod_sofia at 192.168.101.102:5080   RUNNING (0)
>
> =================================================================================================
>
> Thanks in advance
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Fri Mar 19 07:24:40 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 19 Mar 2010 10:24:40 -0400
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <AD5B87549E604DD3BBD73065A1EF6317@cune.pri>
References: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
	<AD5B87549E604DD3BBD73065A1EF6317@cune.pri>
Message-ID: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>

On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann <Russell.Mosemann at cune.org
> wrote:

>
> Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all
> that bad here, but it would be nice to use EC.
>
>
You can always get a Sangoma board with HWEC :-)

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Fri Mar 19 07:30:43 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 19 Mar 2010 10:30:43 -0400
Subject: [Freeswitch-users] openzap no audio?
In-Reply-To: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
References: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
Message-ID: <c4d05cbe1003190730k9bf6802vcc25a362c64d2edd@mail.gmail.com>

Hello,

People have reported problems with audio using DAHDI with software EC. If
you have it enabled, disable it and see if that helps ( you may have echo
though ).

Few ( if any ) of the FreeSWITCH developers use Digium hardware, most use
Sangoma, if you want something that works and is completely supported go
with Sangoma then.

On Thu, Mar 18, 2010 at 1:07 AM, Tom Christensen <pavera at gmail.com> wrote:

> I've got freeswitch installed with a digium te110p hooked up to a PRI.
>
> calls through the system work fine SIP->SIP, calling and receiving calls
> over the PRI works, (IE the signalling part, a call can be made and incoming
> calls come in over the PRI and I can route them based on DID), however,
> there is no audio to either party.  I've checked, and there is an RTP stream
> running while the call is up, there is no reason it would be blocked (IE,
> SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
> few posts that talked about no audio for a few seconds after a call is
> connected, or that calls would connect and then drop after a few seconds.
>  This is different, I can leave the call up for many minutes, it doesn't
> drop, and audio never starts coming through.
>
> I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
> the latest release of zaptel first, freeswitch wouldn't even load with that.
>
> I'm going to try libpri now I think as I've exhausted all the other options
> I can think of.
>
> Any input/ideas would be greatly appreciated.
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Russell.Mosemann at cune.org  Fri Mar 19 07:49:38 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 19 Mar 2010 14:49:38 -0000
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>
Message-ID: <20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>

Moises Silva <moises.silva at gmail.com> said:
> 
> On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann
<Russell.Mosemann at cune.org
> > wrote:
> 
> >
> > Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all
> > that bad here, but it would be nice to use EC.
> >
> >
> You can always get a Sangoma board with HWEC :-)

I'd love to, believe me. In the past, we had four Sangoma data T1 boards
and load balanced our connection to the Internet over them. The hardware
was solid, setup was pretty easy and I got quick answers from support. It
was great.

Unfortunately, there is absolutely no money for the current project. We
had to scrounge a throwaway server and buy a cheap PRI T1 card off of
Ebay. We happened to acquire an additional T1 license on our Siemens
telephone switch as part of another project, and we bought a used T1 card
for the telephone switch off of Ebay. We've only spent about $100 on
something that could easily cost $10,000. We're hoping that after we get
a proof of concept going, the right people will see the value and fund a
better solution. We'll keep you mind. ;-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From lloydie.t at googlemail.com  Fri Mar 19 07:53:10 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Fri, 19 Mar 2010 14:53:10 +0000
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
In-Reply-To: <c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>
References: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
	<c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>
Message-ID: <8b61bd671003190753t506d0c63t6b92c032e3830400@mail.gmail.com>

Thanks Milena. I think I have a much better understanding of how it works,
the main thing being you have to have an entry in the DNS server to point
the multi-tenancy domain name to the IP address of the FS server and make
sure the SIP client also has the correct domain name. All is now better with
the following config (unless you can see a glaring error).
User Directory
------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="User Directory">
<domain name="$${domain}">
<groups>
  <group name="default">
  <users>
  <user id="203">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      <param name="vm-password" value="1234"/>
      <param name="vm-disk-quota" value="30"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
      <variable name="accountcode" value="203"/>
      <variable name="day_bar" value="3"/>
      <variable name="night_bar" value="3"/>
      <variable name="effective_caller_id_name" value="Extension 203"/>
      <variable name="effective_caller_id_number" value="203"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_number}"/>
    </variables>
  </user>
  </users>
 </group>
</groups>
</domain>
</section>
</document>
--------------------------------------------------

Company Directory
------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="Company Directory">
<domain name="$${domain}">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="default"/>
  </variables>
</domain>
<domain name="phitel.phisys.dom">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="phitel.phisys.dom"/>
  </variables>
</domain>
</section>
</document>
--------------------------------------------------------

On 19 March 2010 13:39, Milena <testeador01 at gmail.com> wrote:

> Hello,
>
> None of them are ok for the domain you want to register to, when you set up
> multiple domains, you have to return the results with the domain name you're
> giving information about, make it return another xml with
> <domain name="192.168.101.102">
>
> 2010/3/18 lloyd thomas <lloydie.t at googlemail.com>
>
>> I eventually managed to get mod_xml_curl working for me. Now I trying to
>> be a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
>> wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But
>> I get an error when a phone tries to register.
>>
>> 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
>> 203 at 192.168.101.102]
>> You must define a domain called '192.168.101.102' in your directory and
>> add a user with the id="203" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
>> 192.168.91.28
>>
>> I think I am nearly there,but I suspect there is something wrong with the
>> resulting 'user directory' xml.
>> ---------------------------------------------------------------
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="directory" description="User Directory">
>> <domain name="default">
>> <groups>
>>   <group name="default">
>>   <users>
>>   <user id="203">
>>     <params>
>>       <param name="password" value="$${default_password}"/>
>>       <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>       <param name="vm-password" value="1234"/>
>>       <param name="vm-disk-quota" value="30"/>
>>     </params>
>>     <variables>
>>       <variable name="user_context" value="default"/>
>>       <variable name="accountcode" value="203"/>
>>       <variable name="day_bar" value="3"/>
>>       <variable name="night_bar" value="3"/>
>>       <variable name="effective_caller_id_name" value="Extension 203"/>
>>       <variable name="effective_caller_id_number" value="203"/>
>>       <variable name="outbound_caller_id_name"
>> value="$${outbound_caller_name}"/>
>>       <variable name="outbound_caller_id_number"
>> value="$${outbound_caller_number}"/>
>>     </variables>
>>   </user>
>>   </users>
>>  </group>
>> </groups>
>> </domain>
>> </section>
>> </document>
>> --------------------------------------------------------------------
>>
>> I think the Company Directory XML is OK
>> --------------------------------------------------------------------
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="directory" description="Company Directory">
>> <domain name="default">
>>   <params>
>>     <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>   </params>
>>   <variables>
>>     <variable name="record_stereo" value="true"/>
>>     <variable name="default_gateway" value="$${default_provider}"/>
>>     <variable name="default_areacode" value="$${default_areacode}"/>
>>     <variable name="transfer_fallback_extension" value="operator"/>
>>     <variable name="user_context" value="default"/>
>>   </variables>
>> </domain>
>> <domain name="phisystems.dom">
>>   <params>
>>     <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>   </params>
>>   <variables>
>>     <variable name="record_stereo" value="true"/>
>>     <variable name="default_gateway" value="$${default_provider}"/>
>>     <variable name="default_areacode" value="$${default_areacode}"/>
>>     <variable name="transfer_fallback_extension" value="operator"/>
>>     <variable name="user_context" value="phisystems.dom"/>
>>   </variables>
>> </domain>
>> </section>
>> </document>
>> --------------------------------------------------------------
>>
>> I get the following result when I 'sofia status'
>>                      Name          Type
>> Data     State
>>
>> =================================================================================================
>>                  internal       profile
>> sip:mod_sofia at 192.168.101.102:5060   RUNNING (0)
>>            phisystems.dom         alias
>> internal     ALIASED
>>             internal-ipv6       profile           sip:mod_sofia@[::1]:5060
>> RUNNING (0)
>>                   default         alias
>> internal     ALIASED
>>                  external       profile
>> sip:mod_sofia at 192.168.101.102:5080   RUNNING (0)
>>
>> =================================================================================================
>>
>> Thanks in advance
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Fri Mar 19 08:03:45 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 19 Mar 2010 11:03:45 -0400
Subject: [Freeswitch-users] make errors - mod_cepstral
Message-ID: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>

Hi All,

I am trying to enable flite & cepstral on a existing
FreeSWITCH installation. I follow the following steps

1. build directory  cp modules.con.in modules.conf

2. edit modules.conf
remove the comments from the following

asr_tts/mod_flite
asr_tts/mod_cepstral

3../bootstrap.sh

4. ./configure

5. make

I get the following error

make[5]: *** [mod_cepstral.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1

How can I successfully enable the mod_cepstral and flite.



Thanks
Lloyd
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From brian at freeswitch.org  Fri Mar 19 08:09:46 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 10:09:46 -0500
Subject: [Freeswitch-users] make errors - mod_cepstral
In-Reply-To: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>
References: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>
Message-ID: <A767445F-85A7-42D4-82DC-93561665502E@freeswitch.org>

chances are you don't have cepstral installed which is required to use/compile it.

/b

On Mar 19, 2010, at 10:03 AM, Aloysius Lloyd wrote:

> make[5]: *** [mod_cepstral.lo] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_cepstral-all] Error 1
> make[2]: *** [all-recursive] Error 1



From fs-list at communicatefreely.net  Fri Mar 19 08:12:27 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 11:12:27 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
	<c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>
Message-ID: <4BA3945B.30104@communicatefreely.net>

I think there are advantages, at least from a functionality standpoint.

If you have a cluster setup, the core DB can be shared across FS machines.  This gives each machine
access to things like presence, voice mail counts and settings, and the mod_limit counts can be
global, as opposed to per-machine numbers.  This is handy if you want to enforce a channel count for
 a customer.  You can also share SIP registrations across the cluster.  I haven't tried this too
much yet, but it should be possible for a phone to register to machine A.  If machine B needs to
send a call to the phone, it has the contact data and can send the call to the phone directly (as
long as there are no NAT issues here).  I don't know how often each machine polls the DB for
presence information.  There might be a lag for things like BLF lamps across the cluster, although I
think there is another way to solve that.

You can also access the data in the DB with your PHP scripts for status monitoring and other useful
things.  I have heard that altering this data is a bad idea, but running queries on it can be handy.
   You can make a nice web dashboard that will give you all registered endpoints, current SIP
dialogs, channel counts (via mod_limit), voice mail counts, etc.  By accessing the voice mail
tables, you can also make an XML "visual voicemail" interface for phones like the Aastra 55i and 57i.

Hope that's helpful.

-Tim

Fraser Redmond wrote:
> Is there any advantage at all to putting the core database on your own
> external db? Or is it just a control freak thing? I was looking at this
> the other day, and only ever read that it was possible, not anything
> about why you'd want to do it.
> 
> Cheers,
> Fraser
> 
> 
> 
> 
> On Thu, Mar 18, 2010 at 5:41 PM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>      Its really up to you it works both ways.  But I would recommend a
>     dedicated network for interconnection and management and not run
>     that over the public/sip facing interfaces.
> 
>     /b
> 
>     On Mar 18, 2010, at 12:37 PM, Todd wrote:
> 
>>     Thanks folks!  I believe we will be doing most of our coding in
>>     php.  We are also going to have independent DB servers?. I was
>>     also wondering if putting the FS database on a DB server is a god
>>     idea, or if it should be left on the FS box.  ?Preciate it yall. 
>>     Learning is phun
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Suneel.Papineni at mettoni.com  Fri Mar 19 08:20:06 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 15:20:06 -0000
Subject: [Freeswitch-users] Conference with FSComm
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>

Hi,

 

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

 

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

 

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

 

Thanks & Regards

Suneel

 


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From kenfulmer at icstechnologysolutions.com  Fri Mar 19 08:13:34 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Fri, 19 Mar 2010 10:13:34 -0500
Subject: [Freeswitch-users] Newbie Questions
Message-ID: <00a701cac776$bf147de0$3d3d79a0$@com>

1.       We would like to transcode from g711mu internally to g729 for PSTN
calls. Is this possible? 

2.       We are using sipX as an open source PBX. We'd like to use
FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
phones use SIP REFER messages to do call transfers and our ITSPs can't
handle those. So, we've tried the "under the hood" FreeSwitch within sipX.
It works for many applications but we need to scale a unified system for
larger customer deployments. If we want to create a trunk to an ITSP and a
trunk to an internal PBX, would we use the external gateway XML files for
both? We don't plan to point any internal phones to the FreeSwitch box
(we're using sipX as an internal PBX). 

3.       If we use multiple ITSPs, can we assign a priority to each one, so
one will be chosen first? 

4.       I understand no performance metrics are posted. Can anyone share
their experiences with passing RTP through the server? Does RTP affect
performance in a big way (very open ended question, I know). We are looking
for ball park answers so we'll have some idea of scalability for our
designs. 

 

Thanks,

 

Ken Fulmer

 

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From robert.hadley at teotech.com  Fri Mar 19 09:10:08 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 09:10:08 -0700
Subject: [Freeswitch-users] How does mod_directory get users information?
Message-ID: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>

Hello,

 

Where does mod_directory get the user information it uses to put into the
database in creates?

 

Thanks,

Robert

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From infos at madovsky.org  Fri Mar 19 09:29:10 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 12:29:10 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <F1775BE5441B4234B9A127CD3F6AC2A8@MOBILEE1705>

I set FS internal on port 5080 and external 5060.
2 softphones on public IP, so no nat.
the sip invite is ok, but there's a media timeout..
port 5080 and 5060 are opened on my firewall
must I set any nat params anyway ?

Thanks

Franck
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From fs-list at communicatefreely.net  Fri Mar 19 09:57:14 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 12:57:14 -0400
Subject: [Freeswitch-users] How does mod_directory get users information?
In-Reply-To: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
References: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
Message-ID: <4BA3ACEA.1040503@communicatefreely.net>

It looks in the user directory.

It will request users in the domain specified when you invoke it.

Something I learned (with help from the author), was that the users have to be in a group.  The
default group will do.

If you are using xml_curl to generate your directory dynamically, you have to set your script up so
that it will return all users in the domain when the user isn't specified in the query.  You don't
have to return all the variables though, just the user's name and number will usually work.

Robert Hadley wrote:
> Hello,
> 
>  
> 
> Where does mod_directory get the user information it uses to put into
> the database in creates?
> 
>  
> 
> Thanks,
> 
> Robert
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Fri Mar 19 10:05:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:05:04 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>

with bypass-media=true everything is ok, 
but I need to pass media through FS, I missed
some settings but don't know where...
I use default configs, inverted the internal and external port
and commented out all nat params.

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 12:29 PM
  Subject: rtp timeout issue


  I set FS internal on port 5080 and external 5060.
  2 softphones on public IP, so no nat.
  the sip invite is ok, but there's a media timeout..
  port 5080 and 5060 are opened on my firewall
  must I set any nat params anyway ?

  Thanks

  Franck
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From anthony.minessale at gmail.com  Fri Mar 19 10:17:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:17:17 -0500
Subject: [Freeswitch-users] rtp timeout issue
In-Reply-To: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
Message-ID: <191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>

you should also unblock the rtp port range
16384-32768 udp


On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

>  with bypass-media=true everything is ok,
> but I need to pass media through FS, I missed
> some settings but don't know where...
> I use default configs, inverted the internal and external port
> and commented out all nat params.
>
> Thanks
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 12:29 PM
> *Subject:* rtp timeout issue
>
> I set FS internal on port 5080 and external 5060.
> 2 softphones on public IP, so no nat.
> the sip invite is ok, but there's a media timeout..
> port 5080 and 5060 are opened on my firewall
> must I set any nat params anyway ?
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 19 10:21:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:21:46 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <00a701cac776$bf147de0$3d3d79a0$@com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
Message-ID: <191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>

If you plan to transcode g729 to g711 it will impact performance
dramatically but you should still get a few hundered channels up on a nice
server box.  You could contact consulting at freeswitch.org about g729 license
options.

With no g729 you could scale a nice 8 core box into a few thousand channels
pushing rtp.

Do you have a requirement to transcode g729?  you could just use g711 right
from the phone couldn't you?


On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  1.       We would like to transcode from g711mu internally to g729 for
> PSTN calls. Is this possible?
>
> 2.       We are using sipX as an open source PBX. We?d like to use
> FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
> phones use SIP REFER messages to do call transfers and our ITSPs can?t
> handle those. So, we?ve tried the ?under the hood? FreeSwitch within sipX.
> It works for many applications but we need to scale a unified system for
> larger customer deployments. If we want to create a trunk to an ITSP and a
> trunk to an internal PBX, would we use the external gateway XML files for
> both? We don?t plan to point any internal phones to the FreeSwitch box
> (we?re using sipX as an internal PBX).
>
> 3.       If we use multiple ITSPs, can we assign a priority to each one,
> so one will be chosen first?
>
> 4.       I understand no performance metrics are posted. Can anyone share
> their experiences with passing RTP through the server? Does RTP affect
> performance in a big way (very open ended question, I know). We are looking
> for ball park answers so we?ll have some idea of scalability for our
> designs.
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Fri Mar 19 10:29:35 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:29:35 -0400
Subject: [Freeswitch-users] rtp timeout issue
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
	<191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>
Message-ID: <56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>

yes I did it also.
the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:17 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  you should also unblock the rtp port range
  16384-32768 udp



  On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

    with bypass-media=true everything is ok, 
    but I need to pass media through FS, I missed
    some settings but don't know where...
    I use default configs, inverted the internal and external port
    and commented out all nat params.

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, March 19, 2010 12:29 PM
      Subject: rtp timeout issue


      I set FS internal on port 5080 and external 5060.
      2 softphones on public IP, so no nat.
      the sip invite is ok, but there's a media timeout..
      port 5080 and 5060 are opened on my firewall
      must I set any nat params anyway ?

      Thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

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



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


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From anthony.minessale at gmail.com  Fri Mar 19 10:42:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:42:33 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>
References: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>
	<20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>
Message-ID: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>

Look at it this way.

The PRI signaling is a data channel, it's got nothing to do with echo
cancelers.  The echo can is for the audio legs.
The OSLEC is a kernel module and dahdi is Digiums driver also a kernel
module. OpenZAP only opens these devices and uses them as advertised.

Keep in mind, When we added support for digium cards it was when there was
still zaptel and every day they change things in their code which is mostly
kernel space and they have absolutely no interest in having it work on
FreeSWITCH, even though we are both open source applications they treat
FreeSWITCH like a threat.

You are trying to escape from asterisk but you are still using mostly their
technology.

So bottom line changing around which pri stack you use is not going to solve
audio problems.
All OpenZAP does with digium cards is open the device and read and write
from it.
If you are getting no audio the problem is happening in the kernel before
OpenZAP gets to see anything.


My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
there is a line
echo_cancel_level => 64

try erasing or commenting it out with a leading ;

>From my recollection when you use OSLEC, you have to disable the
dahdi/zaptel echo canceler.

This is the only setting I can think of that allows OpenZAP to influence the
dahdi kernel mod.

BTW, you got that card for $100 because it really is worth that much, maybe
less ;)

Everyone who uses Sangoma cards with OpenZAP have no audio problems at all.




On Fri, Mar 19, 2010 at 9:49 AM, <Russell.Mosemann at cune.org> wrote:

> Moises Silva <moises.silva at gmail.com> said:
> >
> > On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann
> <Russell.Mosemann at cune.org
> > > wrote:
> >
> > >
> > > Sorry, I'm looking for a solution, myself. The echo over the T1 isn't
> all
> > > that bad here, but it would be nice to use EC.
> > >
> > >
> > You can always get a Sangoma board with HWEC :-)
>
> I'd love to, believe me. In the past, we had four Sangoma data T1 boards
> and load balanced our connection to the Internet over them. The hardware
> was solid, setup was pretty easy and I got quick answers from support. It
> was great.
>
> Unfortunately, there is absolutely no money for the current project. We
> had to scrounge a throwaway server and buy a cheap PRI T1 card off of
> Ebay. We happened to acquire an additional T1 license on our Siemens
> telephone switch as part of another project, and we bought a used T1 card
> for the telephone switch off of Ebay. We've only spent about $100 on
> something that could easily cost $10,000. We're hoping that after we get
> a proof of concept going, the right people will see the value and fund a
> better solution. We'll keep you mind. ;-)
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 19 10:44:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:44:09 -0500
Subject: [Freeswitch-users] rtp timeout issue
In-Reply-To: <56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
	<191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>
	<56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
Message-ID: <191c3a031003191044n3334b4ebm5abe7ce52c9e799b@mail.gmail.com>

try stopping iptables for 1 call to make sure it is not related to that.



On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

>  yes I did it also.
> the issue is only noticeable if I use 2 softphones on the same public IP
> where there is FS.
> I tried from another PC (even through nat) and it works fine... Arrgh, I'm
> lost
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 1:17 PM
> *Subject:* Re: [Freeswitch-users] rtp timeout issue
>
> you should also unblock the rtp port range
> 16384-32768 udp
>
>
> On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  with bypass-media=true everything is ok,
>> but I need to pass media through FS, I missed
>> some settings but don't know where...
>> I use default configs, inverted the internal and external port
>> and commented out all nat params.
>>
>> Thanks
>>
>> ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, March 19, 2010 12:29 PM
>> *Subject:* rtp timeout issue
>>
>> I set FS internal on port 5080 and external 5060.
>> 2 softphones on public IP, so no nat.
>> the sip invite is ok, but there's a media timeout..
>> port 5080 and 5060 are opened on my firewall
>> must I set any nat params anyway ?
>>
>> Thanks
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Fri Mar 19 10:53:18 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:53:18 -0400
Subject: [Freeswitch-users] rtp timeout issue
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705><191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com><56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
	<191c3a031003191044n3334b4ebm5abe7ce52c9e799b@mail.gmail.com>
Message-ID: <87DAD62EECE64DF9B33E3089A3D680A1@MOBILEE1705>

bingo.... :), 
grrrr, now I need to read all my iptables :(
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:44 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  try stopping iptables for 1 call to make sure it is not related to that.




  On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

    yes I did it also.
    the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
    I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, March 19, 2010 1:17 PM
      Subject: Re: [Freeswitch-users] rtp timeout issue


      you should also unblock the rtp port range
      16384-32768 udp



      On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

        with bypass-media=true everything is ok, 
        but I need to pass media through FS, I missed
        some settings but don't know where...
        I use default configs, inverted the internal and external port
        and commented out all nat params.

        Thanks
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Friday, March 19, 2010 12:29 PM
          Subject: rtp timeout issue


          I set FS internal on port 5080 and external 5060.
          2 softphones on public IP, so no nat.
          the sip invite is ok, but there's a media timeout..
          port 5080 and 5060 are opened on my firewall
          must I set any nat params anyway ?

          Thanks

          Franck

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





      -- 
      Anthony Minessale II

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

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

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



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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Mar 19 10:58:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:58:01 -0500
Subject: [Freeswitch-users] variables in gateway profile
In-Reply-To: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
Message-ID: <191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>

yes and you can specify which direction you want to set them on, "inbound",
"outbound" or "both"

<variables>
  <variable name="foo" data="bar" direction="inbound"/>
</variables>


On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:

>  Is it possible to create <variable> in gateway profiles like users
> profiles ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Fri Mar 19 11:19:34 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 19 Mar 2010 14:19:34 -0400
Subject: [Freeswitch-users] Originate timeout with multiple destinations
Message-ID: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>

Hi,

I need to do an originate to multiple destinations, but I need that the
first call has a timeout of 17 seconds, and if it's not answered, then call
the other destinations.

I was thinking something like:

originate
{ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx,
sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz


I tried using the call_timeout, timeout and originate_timeout variables in
the square brackets for the first dialstring, but it didn't work. Is there a
way of doing this with 1 originate command?


Thanks!
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From Suneel.Papineni at mettoni.com  Fri Mar 19 11:20:44 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 18:20:44 -0000
Subject: [Freeswitch-users] [Freeswitch-dev] Conference with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>

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

 

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

 

Thanks & Regards

Suneel

 

 

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

 

Hi,

 

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

 

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

 

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

 

Thanks & Regards

Suneel

 

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From Russell.Mosemann at cune.org  Fri Mar 19 11:21:29 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 19 Mar 2010 18:21:29 -0000
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
Message-ID: <20100319182129.49EE5129EA7@cuneorg-email.cune.pri>

Anthony Minessale <anthony.minessale at gmail.com> said:

> All OpenZAP does with digium cards is open the device and read
> and write from it. If you are getting no audio the problem is
> happening in the kernel before OpenZAP gets to see anything.

That provides a starting point for checking things.

> My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
> there is a line
> echo_cancel_level => 64
> 
> try erasing or commenting it out with a leading ;

I did a quick test, and that didn't solve the problem. I'll have to look
into it further.

> >From my recollection when you use OSLEC, you have to disable the
> dahdi/zaptel echo canceler.

That has changed in DAHDI. There is no default echo canceler. The
specific echo canceler (if any) must be specified for each channel (or
range of channels). OSLEC is referenced like any other echo canceler. For
example,

echocanceller=oslec,1-23

> BTW, you got that card for $100 because it really is worth
> that much, maybe less ;)

No arguments there. This is merely a toy system, and audio quality is not
a factor. We just want to demonstrate that this can be useful for us.
Based on past experience with Sangoma products and support, it would be
wonderful to have a card from them.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From infos at madovsky.org  Fri Mar 19 11:30:17 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 14:30:17 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <30060395924C442EBEBD680E3152B9BE@MOBILEE1705>

Found the iptables lines that caused problem :

#### #Prevent external packets from using loopback
-A INPUT -i br0 -s 127.0.0.1 -j DROP
-A FORWARD -i br0 -s 127.0.0.1 -j DROP
-A INPUT -i br0 -d 127.0.0.1 -j DROP
-A FORWARD -i br0 -d 127.0.0.1 -j DROP

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:53 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  bingo.... :), 
  grrrr, now I need to read all my iptables :(
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, March 19, 2010 1:44 PM
    Subject: Re: [Freeswitch-users] rtp timeout issue


    try stopping iptables for 1 call to make sure it is not related to that.




    On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

      yes I did it also.
      the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
      I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Friday, March 19, 2010 1:17 PM
        Subject: Re: [Freeswitch-users] rtp timeout issue


        you should also unblock the rtp port range
        16384-32768 udp



        On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

          with bypass-media=true everything is ok, 
          but I need to pass media through FS, I missed
          some settings but don't know where...
          I use default configs, inverted the internal and external port
          and commented out all nat params.

          Thanks
            ----- Original Message ----- 
            From: Madovsky 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Friday, March 19, 2010 12:29 PM
            Subject: rtp timeout issue


            I set FS internal on port 5080 and external 5060.
            2 softphones on public IP, so no nat.
            the sip invite is ok, but there's a media timeout..
            port 5080 and 5060 are opened on my firewall
            must I set any nat params anyway ?

            Thanks

            Franck

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        -- 
        Anthony Minessale II

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

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
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    -- 
    Anthony Minessale II

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

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
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From robert.hadley at teotech.com  Fri Mar 19 11:30:55 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 11:30:55 -0700
Subject: [Freeswitch-users] How does mod_directory get users information?
In-Reply-To: <4BA3ACEA.1040503@communicatefreely.net>
References: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
	<4BA3ACEA.1040503@communicatefreely.net>
Message-ID: <26B2C7A0869F40CDBBA7457B31FF9B75@greyhawk.tonecommander.com>

Thanks Tim,

It's working now by setting the "effective_caller_id_name" variables to
actual names.

Question: What is the way to adjust the sound level of the directory
phrases, especially the TTS portions?

The playback of the .wav files sound loud enough, but I can barely hear to
spelled names.

Thanks,
Robert


-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net] 
Sent: Friday, March 19, 2010 9:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How does mod_directory get users
information?

It looks in the user directory.

It will request users in the domain specified when you invoke it.

Something I learned (with help from the author), was that the users have to
be in a group.  The
default group will do.

If you are using xml_curl to generate your directory dynamically, you have
to set your script up so
that it will return all users in the domain when the user isn't specified in
the query.  You don't
have to return all the variables though, just the user's name and number
will usually work.

Robert Hadley wrote:
> Hello,
> 
>  
> 
> Where does mod_directory get the user information it uses to put into
> the database in creates?
> 
>  
> 
> Thanks,
> 
> Robert
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From fraserredmond at gmail.com  Fri Mar 19 11:43:28 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 19 Mar 2010 18:43:28 +0000
Subject: [Freeswitch-users] Open source softphone that can handle g729
Message-ID: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>

We're looking for an open source softphones that can handle g729 that we can
extend with extra functionality (connecting to other applications, and
things on the intranet.)

Does anyone have any they'd suggest?

Brian West suggested FSComm with the qualifier "it still needs more work" -
has anyone used it much (especially in windows), or know what it's
development status is? (Looking in the source browser jmesquita spent a lot
of time on it in January, but not much since.)

Cheers,
Fraser
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From brian at freeswitch.org  Fri Mar 19 11:48:32 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 13:48:32 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <85353439-E829-4907-8B74-F877D80BCFB3@freeswitch.org>

Someone that really needs it such as yourself might think about helping the project along with some $$ ;)

FSComm will be able to use the same G729 that FreeSWITCH uses once its officially launched.

/b

On Mar 19, 2010, at 1:43 PM, Fraser Redmond wrote:

> We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.)
> 
> Does anyone have any they'd suggest?
> 
> Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.)
> 
> Cheers,
> Fraser



From anthony.minessale at gmail.com  Fri Mar 19 12:00:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:00:27 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
Message-ID: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>

I had a look at the code and I discovered that the zt mod defaults to a echo
can level of 64 internally.
I did a patch to the code that should allow it to be explicitly off when you
omit that param.

If you could update to latest trunk and test it, it should be a real test of
my theory.

If you still have the problem the next step would be to add some debugging
code to
the ozmod_zt read and write functions to see if it's sending data up to the
app.

Maybe there are some new ioctls or something they added for interop with
OSLEC that we don't know about.



On Fri, Mar 19, 2010 at 1:21 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > All OpenZAP does with digium cards is open the device and read
> > and write from it. If you are getting no audio the problem is
> > happening in the kernel before OpenZAP gets to see anything.
>
> That provides a starting point for checking things.
>
> > My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
> > there is a line
> > echo_cancel_level => 64
> >
> > try erasing or commenting it out with a leading ;
>
> I did a quick test, and that didn't solve the problem. I'll have to look
> into it further.
>
> > >From my recollection when you use OSLEC, you have to disable the
> > dahdi/zaptel echo canceler.
>
> That has changed in DAHDI. There is no default echo canceler. The
> specific echo canceler (if any) must be specified for each channel (or
> range of channels). OSLEC is referenced like any other echo canceler. For
> example,
>
> echocanceller=oslec,1-23
>
> > BTW, you got that card for $100 because it really is worth
> > that much, maybe less ;)
>
> No arguments there. This is merely a toy system, and audio quality is not
> a factor. We just want to demonstrate that this can be useful for us.
> Based on past experience with Sangoma products and support, it would be
> wonderful to have a card from them.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar 19 12:03:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:03:10 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Conference with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
	<3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>
Message-ID: <191c3a031003191203q4448d227gdfd36a0f23120243@mail.gmail.com>

uuid_transfer the uuid of the holding leg first
then uuid_transfer -both on the A leg of the bridged call to

uuid_transfer <holding uuid> conference:<conf name here>@default inline

uuid_transfer -both <a-leg uuid> conference:<conf name here>@default inline


On Fri, Mar 19, 2010 at 1:20 PM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  I am not sure if the following method works. Please let me know how could
> I implement this through event sockets.
>
>
>
> ?A? calls FSComm and FSComm calls ?B?. Now FSComm is having call details of
> all call legs. If possible FSComm now creates a conference and join all 3
> call legs to the conference. This might solve the problem, but could you
> please let me know how can create a conference and join all the legs. Please
> let me know the commands for this. I checked with ?mod_conference? module,
> unable to find respective commands. I tried ?api conference....?, it is
> saying command not found. I tried ?conference.....? command, it is also not
> working.
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Suneel
> Papineni
> *Sent:* 19 March 2010 15:20
> *To:* freeswitch-dev at lists.freeswitch.org
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-dev] Conference with FSComm
>
>
>
> Hi,
>
>
>
> I am trying to achieve 3-way (or 3-party) conference scenario using FSComm.
> Scenario is as follows:
>
>
>
> An incoming call comes to FSComm from an extension ?A? and is answered. Now
> FSComm makes a call to another extension ?B? and is answered. Then I want to
> bridge all three calls and be in conference. So all three parties A, B and
> FSComm will be in conference.
>
>
>
> Could someone please give me an idea how can I achieve this scenario using
> Event Sockets. (want to test with fs_cli using commands).
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
> *************************************************************************
>
> Please consider the environment before printing this e-mail
>
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>
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>
> are addressed. If you have received this email in error please notify
>
> the system manager.  http://www.mettoni.com
>
>
>
> Mettoni Ltd
>
> Registered in England and Wales: 4485956
>
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>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar 19 12:13:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:13:24 -0500
Subject: [Freeswitch-users] Originate timeout with multiple destinations
In-Reply-To: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
References: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
Message-ID: <191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>

leg_timeout is the one for the []


On Fri, Mar 19, 2010 at 1:19 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi,
>
> I need to do an originate to multiple destinations, but I need that the
> first call has a timeout of 17 seconds, and if it's not answered, then call
> the other destinations.
>
> I was thinking something like:
>
> originate
> {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz
>
>
> I tried using the call_timeout, timeout and originate_timeout variables in
> the square brackets for the first dialstring, but it didn't work. Is there a
> way of doing this with 1 originate command?
>
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From infos at madovsky.org  Fri Mar 19 12:21:43 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 15:21:43 -0400
Subject: [Freeswitch-users] variables in gateway profile
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
Message-ID: <E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>

Ok thanks.
but if I have the same var name with different value in 2 different gateways ?

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:58 PM
  Subject: Re: [Freeswitch-users] variables in gateway profile


  yes and you can specify which direction you want to set them on, "inbound", "outbound" or "both"

  <variables>
    <variable name="foo" data="bar" direction="inbound"/>
  </variables>



  On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to create <variable> in gateway profiles like users profiles ?

    Thanks

    Franck

    _______________________________________________
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  -- 
  Anthony Minessale II

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

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

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



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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Fri Mar 19 12:49:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:49:15 -0500
Subject: [Freeswitch-users] variables in gateway profile
In-Reply-To: <E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
	<E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>
Message-ID: <191c3a031003191249g10be7fbbl20023e3e8584e85b@mail.gmail.com>

yes that's ok it will get set based on which one you call


On Fri, Mar 19, 2010 at 2:21 PM, Madovsky <infos at madovsky.org> wrote:

>  Ok thanks.
> but if I have the same var name with different value in 2 different
> gateways ?
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 1:58 PM
> *Subject:* Re: [Freeswitch-users] variables in gateway profile
>
> yes and you can specify which direction you want to set them on, "inbound",
> "outbound" or "both"
>
> <variables>
>   <variable name="foo" data="bar" direction="inbound"/>
> </variables>
>
>
> On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Is it possible to create <variable> in gateway profiles like users
>> profiles ?
>>
>> Thanks
>>
>> Franck
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Fri Mar 19 13:05:22 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 19 Mar 2010 16:05:22 -0400
Subject: [Freeswitch-users] Originate timeout with multiple destinations
In-Reply-To: <191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>
References: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
	<191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>
Message-ID: <1b46b4e81003191305y1ea80c49j27832dd1e6a488a4@mail.gmail.com>

Great, thanks!


On Fri, Mar 19, 2010 at 3:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> leg_timeout is the one for the []
>
>
> On Fri, Mar 19, 2010 at 1:19 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Hi,
>>
>> I need to do an originate to multiple destinations, but I need that the
>> first call has a timeout of 17 seconds, and if it's not answered, then call
>> the other destinations.
>>
>> I was thinking something like:
>>
>> originate
>> {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz
>>
>>
>> I tried using the call_timeout, timeout and originate_timeout variables in
>> the square brackets for the first dialstring, but it didn't work. Is there a
>> way of doing this with 1 originate command?
>>
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Mar 19 13:32:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 19 Mar 2010 15:32:47 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com><20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
Message-ID: <FD3C3036FB7A431D8AB83B100CCC6531@cune.pri>

Anthony Minessale wrote:
> I had a look at the code and I discovered that the zt mod defaults to a
> echo can level of 64 internally. I did a patch to the code that should
> allow it to be explicitly off when you omit that param.

I think that worked. On the softphone side I can hear the caller, now, but they can't hear me. I'm not at my regular computer, and I can't tell if the microphone isn't working, or if it is something else. As soon as get back, I will give it a try. Thanks!

--
Russell Mosemann



From robert.hadley at teotech.com  Fri Mar 19 13:42:56 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 13:42:56 -0700
Subject: [Freeswitch-users] Increase volume of "say" phrases to match
	play-file?
Message-ID: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>

Hello,

 

Is there a way to increase the volume of the "say" phrases to match the
"play-file" phrases?

 

I am using mod_directory, the sound files from
http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/, and
the built-in phrases from /usr/local/freeswitch/conf/lang/en/dir/sounds.xml.

 

 

Thanks,

Robert

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From kenfulmer at icstechnologysolutions.com  Fri Mar 19 14:08:11 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Fri, 19 Mar 2010 16:08:11 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
	<191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>
Message-ID: <016f01cac7a8$48f13180$dad39480$@com>

Thanks for your reply.

 

If we only want to route calls from one provider to another or from the PSTN
to an internal PBX (and vice versa), would we use the external SIP profiles
for each "trunk"? 

 

Thanks,

 

Ken Fulmer

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, March 19, 2010 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

 

If you plan to transcode g729 to g711 it will impact performance
dramatically but you should still get a few hundered channels up on a nice
server box.  You could contact consulting at freeswitch.org about g729 license
options.

With no g729 you could scale a nice 8 core box into a few thousand channels
pushing rtp.

Do you have a requirement to transcode g729?  you could just use g711 right
from the phone couldn't you?
 

On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

1.       We would like to transcode from g711mu internally to g729 for PSTN
calls. Is this possible? 

2.       We are using sipX as an open source PBX. We'd like to use
FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
phones use SIP REFER messages to do call transfers and our ITSPs can't
handle those. So, we've tried the "under the hood" FreeSwitch within sipX.
It works for many applications but we need to scale a unified system for
larger customer deployments. If we want to create a trunk to an ITSP and a
trunk to an internal PBX, would we use the external gateway XML files for
both? We don't plan to point any internal phones to the FreeSwitch box
(we're using sipX as an internal PBX). 

3.       If we use multiple ITSPs, can we assign a priority to each one, so
one will be chosen first? 

4.       I understand no performance metrics are posted. Can anyone share
their experiences with passing RTP through the server? Does RTP affect
performance in a big way (very open ended question, I know). We are looking
for ball park answers so we'll have some idea of scalability for our
designs. 

 

Thanks,

 

Ken Fulmer

 


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




-- 
Anthony Minessale II

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

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

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

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From will.traenkle at yahoo.com  Fri Mar 19 14:24:06 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Mar 2010 14:24:06 -0700 (PDT)
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
Message-ID: <43380.74634.qm@web57607.mail.re1.yahoo.com>

Where/how does mod_directory associate a name with an extension?  Meaning, what variable and where does it get the "last name" of the extension user?

I searched the wiki and could not find much on mod_directory and how it works. Will this information be updated soon?

Thanks,

-Will




________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com> wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?
>
>I heard this was a feature that was implemented or going to be implemented in freeswitch.
>
>If it has been implemented, would you send me info on how to configure it?
>
>If it has not been implemented, would you state when it is planned to be implemented?
>
>I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report back.
-MC


      
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From fs-list at communicatefreely.net  Fri Mar 19 14:39:55 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 17:39:55 -0400
Subject: [Freeswitch-users] mod_voicemail modifications
Message-ID: <4BA3EF2B.5060407@communicatefreely.net>

Anybody out there know the inner workings of the voice mail module?

It looks like I may have to modify the source code to make it do what I want.  It's so close, but I
can't quite get it to work.

I need to change it so that the voicemail_greeting_number channel variable is used to set the
greeting path, but is NOT stored back in the sql database.

I would also like to change it so that choosing the option to e-mail the message does not mark it as
deleted.  I imagine I will get many angry phone calls if I put it into production this way.  The
prompt doesn't tell you that the message will be deleted if you e-mail it, and not all of my users
have e-mail set up correctly.  I would prefer messages not to be deleted unless delete is explicitly
pressed.

I'm pretty new to programming, but I would like to give it a try (unless anyone else out there knows
a better way).

Can someone point me in the right direction?  I mostly want to know where the sql gets updated when
the variable is set.  That wasn't obvious when I looked through the code.

In terms of the e-mail causing the deletion, it looks like the e-mail routine is run in-line with
the delete routine, so I'm not sure if they can be separated easily.

Thanks for any help anyone can provide.

-Tim


From fs-list at communicatefreely.net  Fri Mar 19 14:42:46 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 17:42:46 -0400
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <43380.74634.qm@web57607.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
	<43380.74634.qm@web57607.mail.re1.yahoo.com>
Message-ID: <4BA3EFD6.5030506@communicatefreely.net>

I suppose I could try to post something, now that I have it mostly figured out.  I did get it
working, even with our 100% xml_curl driven directory.

Give me a few days.

-Tim

William Traenkle wrote:
> Where/how does mod_directory associate a name with an extension? 
> Meaning, what variable and where does it get the "last name" of the
> extension user?
> 
> I searched the wiki and could not find much on mod_directory and how it
> works. Will this information be updated soon?
> 
> Thanks,
> 
> -Will
> 
> ------------------------------------------------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, March 18, 2010 11:28:44 AM
> *Subject:* Re: [Freeswitch-users] dial by name directory // mod_directory?
> 
> 
> 
> On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle
> <will.traenkle at yahoo.com <mailto:will.traenkle at yahoo.com>> wrote:
> 
>     Has freeswitch implemented the "Dial by Name" feature where a caller
>     can enter the first x number of characters of an extensions first or
>     last name and have the operator either connect the user to the
>     result and/or state the result's extension number?
> 
>     I heard this was a feature that was implemented or going to be
>     implemented in freeswitch.
> 
>     If it has been implemented, would you send me info on how to
>     configure it?
> 
>     If it has not been implemented, would you state when it is planned
>     to be implemented?
> 
>     I believe the module name is mod_directory, is that correct?
> 
> It is mod_directory. I haven't tried it yet. Please test drive it and
> report back.
> -MC
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From robert.hadley at teotech.com  Fri Mar 19 15:07:48 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 15:07:48 -0700
Subject: [Freeswitch-users] mod_voicemail modifications
In-Reply-To: <4BA3EF2B.5060407@communicatefreely.net>
References: <4BA3EF2B.5060407@communicatefreely.net>
Message-ID: <2DA362E3E39C405DA8557FE7ADA570F7@greyhawk.tonecommander.com>

Hi Tim,

I wrote up a Jira bug for the email key always deleting voicemails.  Andrew
T. responded with a patch that is supposed to fix half of the problem such
that voicemails are not deleted when emailed.  However he didn't include
using the vm-keep-local-after-email setting in the user profile.  I haven't
had a chance to try this patch yet.

See http://jira.freeswitch.org/browse/MODAPP-403

I would be interested in any voicemail fixes being made too.

Regards,
Robert



-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net] 
Sent: Friday, March 19, 2010 2:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_voicemail modifications

Anybody out there know the inner workings of the voice mail module?

It looks like I may have to modify the source code to make it do what I
want.  It's so close, but I
can't quite get it to work.

I need to change it so that the voicemail_greeting_number channel variable
is used to set the
greeting path, but is NOT stored back in the sql database.

I would also like to change it so that choosing the option to e-mail the
message does not mark it as
deleted.  I imagine I will get many angry phone calls if I put it into
production this way.  The
prompt doesn't tell you that the message will be deleted if you e-mail it,
and not all of my users
have e-mail set up correctly.  I would prefer messages not to be deleted
unless delete is explicitly
pressed.

I'm pretty new to programming, but I would like to give it a try (unless
anyone else out there knows
a better way).

Can someone point me in the right direction?  I mostly want to know where
the sql gets updated when
the variable is set.  That wasn't obvious when I looked through the code.

In terms of the e-mail causing the deletion, it looks like the e-mail
routine is run in-line with
the delete routine, so I'm not sure if they can be separated easily.

Thanks for any help anyone can provide.

-Tim





From srinivas.ksvreddy at gmail.com  Fri Mar 19 15:09:22 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 20 Mar 2010 03:39:22 +0530
Subject: [Freeswitch-users] registering with auth id
Message-ID: <f8af5741003191509q251ff12bt6941e118d8b7ea57@mail.gmail.com>

HI,

i have a user 1002 in freeswitch, i am trying to register 2001 with help of
1002, mean with auth id is 1002 for me,
is it possible to register in this way

-- 
Srinivasula Reddy K
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From robert.hadley at teotech.com  Fri Mar 19 15:11:16 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 15:11:16 -0700
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <43380.74634.qm@web57607.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com><87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
	<43380.74634.qm@web57607.mail.re1.yahoo.com>
Message-ID: <051753A8EA1B44A18831F4C0AF63CD4A@greyhawk.tonecommander.com>

Hi Will,

 

One way: Set the effective_caller_id_name in the LineId.xml files in the
profiles you are using.

<include>

  <user id="1045">

    <params>

    </params>

    <variables>

      <variable name="effective_caller_id_name" value="FirstName LastName"/>

 

 

I had asked the same question earlier today.  I have attached the answer I
got.

 

Regards,

Robert

 

 

  _____  

From: William Traenkle [mailto:will.traenkle at yahoo.com] 
Sent: Friday, March 19, 2010 2:24 PM
To: freeswitch-users at lists.freeswitch.org; msc at freeswitch.org
Subject: [Freeswitch-users] mod_directory question

 

Where/how does mod_directory associate a name with an extension?  Meaning,
what variable and where does it get the "last name" of the extension user?

I searched the wiki and could not find much on mod_directory and how it
works. Will this information be updated soon?

Thanks,

-Will

 

  _____  

From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com>
wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can
enter the first x number of characters of an extensions first or last name
and have the operator either connect the user to the result and/or state the
result's extension number?

I heard this was a feature that was implemented or going to be implemented
in freeswitch.

If it has been implemented, would you send me info on how to configure it?

If it has not been implemented, would you state when it is planned to be
implemented?

I believe the module name is mod_directory, is that correct?

It is mod_directory. I haven't tried it yet. Please test drive it and report
back.
-MC

 

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From: "Tim St. Pierre" <fs-list at communicatefreely.net>
Subject: Re: [Freeswitch-users] How does mod_directory get users information?
Date: Fri, 19 Mar 2010 09:57:14 -0700
Size: 4755
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100319/0c681242/attachment.mht 

From msc at freeswitch.org  Fri Mar 19 16:27:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Mar 2010 16:27:11 -0700
Subject: [Freeswitch-users] Increase volume of "say" phrases to match
	play-file?
In-Reply-To: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>
References: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91003191627u7d2ce3e6j74f2241e4a8b99f2@mail.gmail.com>

If these are all static sound files then you should just normalize the ones
that are too loud. Brian uses this to normalize our stock english sound
files:

http://svn.freeswitch.org/svn/sounds/trunk/en/us/callie/dist.sh

-MC

On Fri, Mar 19, 2010 at 1:42 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  Hello,
>
>
>
> Is there a way to increase the volume of the ?say? phrases to match the
> ?play-file? phrases?
>
>
>
> I am using mod_directory, the sound files from
> http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/,
> and the built-in phrases from
> /usr/local/freeswitch/conf/lang/en/dir/sounds.xml.
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Fri Mar 19 17:26:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Mar 2010 11:26:31 +1100
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>
References: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
	<20100318132319.2A5BA312875@cuneorg-email.cune.pri>
	<c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>
Message-ID: <3c5cf5261003191726n4acc3bc0l306e38e6f0da4bde@mail.gmail.com>

On 19 March 2010 01:19, Moises Silva <moises.silva at gmail.com> wrote:
> do svn update ( that will fix some of the errors but NOT the critical one).
> The critical one is "failure configuring device ... " and I think that one
> is because you did not run "dahdi_cfg" utility.

Looks like the main part of the error was my fault. I hadn't realized
my distribution (Voyage) was missing the user space tools for dahdi
(only had the kernel modules).

Now fixed.

I still do get some errors when reloading mod_openzap, even with the
latest subversion update of freeswitch:

2010-03-20 11:24:43.081334 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.081334 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.094327 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.094327 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.094327 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.094327 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.532271 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.532271 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.

Not sure if this is my fault or not.

Fortunately it seems to work fine regardless.
-- 
Brian May <brian at microcomaustralia.com.au>


From anthony.minessale at gmail.com  Fri Mar 19 18:36:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 20:36:41 -0500
Subject: [Freeswitch-users] mod_voicemail modifications
In-Reply-To: <191c3a031003191833g32da5897vdcad887008bd0dbe@mail.gmail.com>
References: <4BA3EF2B.5060407@communicatefreely.net>
	<2DA362E3E39C405DA8557FE7ADA570F7@greyhawk.tonecommander.com>
	<191c3a031003191833g32da5897vdcad887008bd0dbe@mail.gmail.com>
Message-ID: <191c3a031003191836t628ecb28u156aeddd0ddcb46b@mail.gmail.com>

I think I have said at least once.
If you have that email delete problem
Update to trunc and Issue make vm-sync

You have an out-of-date vm template in your config.
Once you have the correct template it will not even offer the email option
when you don't have email.

For the other reqs file a bounty or feature req jira for the desired
options.

On Mar 19, 2010 5:13 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:

Hi Tim,

I wrote up a Jira bug for the email key always deleting voicemails.  Andrew
T. responded with a patch that is supposed to fix half of the problem such
that voicemails are not deleted when emailed.  However he didn't include
using the vm-keep-local-after-email setting in the user profile.  I haven't
had a chance to try this patch yet.

See http://jira.freeswitch.org/browse/MODAPP-403

I would be interested in any voicemail fixes being made too.

Regards,
Robert




-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net]
Sent: Fri...
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From sharad at coraltele.com  Fri Mar 19 22:41:42 2010
From: sharad at coraltele.com (sharad)
Date: Sat, 20 Mar 2010 11:11:42 +0530
Subject: [Freeswitch-users] How to find the number of live calls for a
	specific dialplan
Message-ID: <B91AFBDD83964B558D84B4D859794C1F@sharad>

Hi All

If there any way to find out the number of live calls for a specific dialplan.

Regards
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From tayeb.meftah at gmail.com  Sat Mar 20 03:27:16 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 20 Mar 2010 11:27:16 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <4BA4A304.9090401@gmail.com>

hi,
purchase the freeswitch G729 license and use it with FSComm
or develope your own freeswitch based softphone with G729 license from 
brian K West
Le 19/03/2010 19:43, Fraser Redmond a ?crit :
> We're looking for an open source softphones that can handle g729 that 
> we can extend with extra functionality (connecting to other 
> applications, and things on the intranet.)
>
> Does anyone have any they'd suggest?
>
> Brian West suggested FSComm with the qualifier "it still needs more 
> work" - has anyone used it much (especially in windows), or know what 
> it's development status is? (Looking in the source browser jmesquita 
> spent a lot of time on it in January, but not much since.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From raffaele.p.guidi at gmail.com  Sat Mar 20 06:24:04 2010
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 20 Mar 2010 14:24:04 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>

I believe sip communicator does handle g729. It has a strong community and a
plugin model (based on osgi) that makes it easily (ok, kinda...)
customizable. Of course the FSComm/givesomemoneytotheproject option is the
best one. In any case:
http://sip-communicator.org/

<http://sip-communicator.org/>Ciao,
    R

On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> We're looking for an open source softphones that can handle g729 that we
> can extend with extra functionality (connecting to other applications, and
> things on the intranet.)
>
> Does anyone have any they'd suggest?
>
> Brian West suggested FSComm with the qualifier "it still needs more work" -
> has anyone used it much (especially in windows), or know what it's
> development status is? (Looking in the source browser jmesquita spent a lot
> of time on it in January, but not much since.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Sat Mar 20 06:53:24 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 20 Mar 2010 21:53:24 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <4BA4D354.3080009@coppice.org>

On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> I believe sip communicator does handle g729. It has a strong community 
> and a plugin model (based on osgi) that makes it easily (ok, kinda...) 
> customizable. Of course the FSComm/givesomemoneytotheproject option is 
> the best one. In any case:
> http://sip-communicator.org/
>
> Ciao,
>     R
It seems one of their summer of code projects is to implement a G.729 
codec, so it doesn't seem like they have one.
>
> On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond 
> <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
>
>     We're looking for an open source softphones that can handle g729
>     that we can extend with extra functionality (connecting to other
>     applications, and things on the intranet.)
>
>     Does anyone have any they'd suggest?
>
>     Brian West suggested FSComm with the qualifier "it still needs
>     more work" - has anyone used it much (especially in windows), or
>     know what it's development status is? (Looking in the source
>     browser jmesquita spent a lot of time on it in January, but not
>     much since.)
>

Steve



From fraserredmond at gmail.com  Sat Mar 20 08:42:44 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 20 Mar 2010 15:42:44 +0000
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com> 
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com> 
	<4BA4D354.3080009@coppice.org>
Message-ID: <c32ee1cf1003200842u5dfc14f2m918f3477a0d5e7e3@mail.gmail.com>

Thanks, all good info.

FSComm still seems like it needs a lot more work before we could use it, by
which time we'd be settled on something else. Not that that rules out giving
money to the project :-)

I'll take a better look at sip-communicator, the other one we're looking at
is Qute Com - has anyone used that at all?

Cheers,
Fraser




On Sat, Mar 20, 2010 at 1:53 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> > I believe sip communicator does handle g729. It has a strong community
> > and a plugin model (based on osgi) that makes it easily (ok, kinda...)
> > customizable. Of course the FSComm/givesomemoneytotheproject option is
> > the best one. In any case:
> > http://sip-communicator.org/
> >
> > Ciao,
> >     R
> It seems one of their summer of code projects is to implement a G.729
> codec, so it doesn't seem like they have one.
> >
> > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
> > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
> >
> >     We're looking for an open source softphones that can handle g729
> >     that we can extend with extra functionality (connecting to other
> >     applications, and things on the intranet.)
> >
> >     Does anyone have any they'd suggest?
> >
> >     Brian West suggested FSComm with the qualifier "it still needs
> >     more work" - has anyone used it much (especially in windows), or
> >     know what it's development status is? (Looking in the source
> >     browser jmesquita spent a lot of time on it in January, but not
> >     much since.)
> >
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Mar 20 09:02:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 11:02:15 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <E83352F5-E954-421C-81DC-1F621E54DEDD@freeswitch.org>

sip communicator is crap.  Its trying to do way too much and does not a single thing very well at all.  If they have G729 its not legal to use.

/b

On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:

> I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
> http://sip-communicator.org/
> 
> Ciao,
>     R

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From brian at freeswitch.org  Sat Mar 20 09:02:37 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 11:02:37 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<4BA4D354.3080009@coppice.org>
Message-ID: <F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>

I haven't see many of the SOC projects ever produce anything useful as of late.

/b

On Mar 20, 2010, at 8:53 AM, Steve Underwood wrote:

> It seems one of their summer of code projects is to implement a G.729 
> codec, so it doesn't seem like they have one.



From anthony.minessale at gmail.com  Sat Mar 20 09:12:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 20 Mar 2010 11:12:39 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
Message-ID: <191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>

I think the problem is that fs sets the stack size for the core and new
threads to 240k so when the child is born from system() the new process
inherits the small stack size and sendmail needs more than 240k

If you are root, it has the priveledge to raise the stack size to the
default size of 8m and sets it back down when the command completes.

If you are not root you lack these privs and the stack size remains small.

This has been a long time problem how to make a solution that is cross
platform and will not break working configurations.

On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:

 I tried also with sudo and freeswitch user in sudoers with the same result,
segmentation fault.
Anyway my patch works perfectly now, even with sendmail (the only difference
without patch is the "From" is taken from the voicemail.tpl and not the user
variable)


>
> ----- Original Message -----
> From: Fran?ois Legal
> To: freeswitch-users at lists.freeswitch.org


> Sent: Friday, March 19, 2010 6:51 AM
> Subject: Re: [Freeswitch-users] voicemail options [tempora...


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

...
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From infos at madovsky.org  Sat Mar 20 09:42:55 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:42:55 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com><d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<E83352F5-E954-421C-81DC-1F621E54DEDD@freeswitch.org>
Message-ID: <192F31E8CC78429BBE62051BBCDB9A81@MOBILEE1705>

There's linphone, maybe you can add G729 plugin
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:02 PM
  Subject: Re: [Freeswitch-users] Open source softphone that can handle g729


  sip communicator is crap.  Its trying to do way too much and does not a single thing very well at all.  If they have G729 its not legal to use.


  /b


  On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:


    I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
    http://sip-communicator.org/


    Ciao,
        R




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar 20 09:47:37 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:47:37 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705><e2459a682c4993206a5701120a82ed33@thom.fr.eu.org><49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
Message-ID: <F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>

Ok I undertand.
buy Why to continue to use pipe and STDIN since to use "filename" path works ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:12 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  I think the problem is that fs sets the stack size for the core and new threads to 240k so when the child is born from system() the new process inherits the small stack size and sendmail needs more than 240k

  If you are root, it has the priveledge to raise the stack size to the default size of 8m and sets it back down when the command completes.

  If you are not root you lack these privs and the stack size remains small.

  This has been a long time problem how to make a solution that is cross platform and will not break working configurations. 


    On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:


    I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
    Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)

      >
      > ----- Original Message -----
      > From: Fran?ois Legal
      > To: freeswitch-users at lists.freeswitch.org


      > Sent: Friday, March 19, 2010 6:51 AM
      > Subject: Re: [Freeswitch-users] voicemail options [tempora...


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

    ...



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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar 20 09:56:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:56:05 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <C222121AFFAD4392AB636DAB2F091DD5@MOBILEE1705>

Also

http://www.antisip.com/as/index.php

they have a little softphone that manages a lot of codec (G729 included)

Regards
  ----- Original Message ----- 
  From: Raffaele P. Guidi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 9:24 AM
  Subject: Re: [Freeswitch-users] Open source softphone that can handle g729


  I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
  http://sip-communicator.org/


  Ciao,
      R


  On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:

    We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.)

    Does anyone have any they'd suggest?

    Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.)

    Cheers,
    Fraser



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






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From anthony.minessale at gmail.com  Sat Mar 20 09:58:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 20 Mar 2010 11:58:43 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
	<F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
Message-ID: <191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>

I am willing to change anything we have to as long as it works on all
platforms and we can give proper warnings to others.
Maybe we will do something like another param to say if your mailer app
needs a pipe or can accept the filename so we can do both
and leave the defaults to the current way.

Maybe we can also have another option to make a simple relay code internally
that can deliver the file to a mail relay if you set that option.



On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:

>  Ok I undertand.
> buy Why to continue to use pipe and STDIN since to use "filename" path
> works ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Saturday, March 20, 2010 12:12 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> I think the problem is that fs sets the stack size for the core and new
> threads to 240k so when the child is born from system() the new process
> inherits the small stack size and sendmail needs more than 240k
>
> If you are root, it has the priveledge to raise the stack size to the
> default size of 8m and sets it back down when the command completes.
>
> If you are not root you lack these privs and the stack size remains small.
>
> This has been a long time problem how to make a solution that is cross
> platform and will not break working configurations.
>
> On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:
>
>  I tried also with sudo and freeswitch user in sudoers with the same
> result, segmentation fault.
> Anyway my patch works perfectly now, even with sendmail (the only
> difference without patch is the "From" is taken from the voicemail.tpl and
> not the user variable)
>
>
> >
> > ----- Original Message -----
> > From: Fran?ois Legal
> > To: freeswitch-users at lists.freeswitch.org
>
>
> > Sent: Friday, March 19, 2010 6:51 AM
> > Subject: Re: [Freeswitch-users] voicemail options [tempora...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ...
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Sat Mar 20 10:13:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 13:13:10 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705><e2459a682c4993206a5701120a82ed33@thom.fr.eu.org><49B3842E181347948484F22AB2FABBD0@MOBILEE1705><191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com><F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
Message-ID: <A643303450404497B14F79175BEC0EF4@MOBILEE1705>

ok irght.
So for now I will continue to use my patch to make it work.
I will give to you the status of any system upgrade that resolves the problem
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:58 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  I am willing to change anything we have to as long as it works on all platforms and we can give proper warnings to others.
  Maybe we will do something like another param to say if your mailer app needs a pipe or can accept the filename so we can do both
  and leave the defaults to the current way.

  Maybe we can also have another option to make a simple relay code internally that can deliver the file to a mail relay if you set that option.




  On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:

    Ok I undertand.
    buy Why to continue to use pipe and STDIN since to use "filename" path works ?
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Saturday, March 20, 2010 12:12 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      I think the problem is that fs sets the stack size for the core and new threads to 240k so when the child is born from system() the new process inherits the small stack size and sendmail needs more than 240k

      If you are root, it has the priveledge to raise the stack size to the default size of 8m and sets it back down when the command completes.

      If you are not root you lack these privs and the stack size remains small.

      This has been a long time problem how to make a solution that is cross platform and will not break working configurations. 


        On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:


        I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
        Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)

          >
          > ----- Original Message -----
          > From: Fran?ois Legal
          > To: freeswitch-users at lists.freeswitch.org


          > Sent: Friday, March 19, 2010 6:51 AM
          > Subject: Re: [Freeswitch-users] voicemail options [tempora...


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

        ...



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





  -- 
  Anthony Minessale II

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

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

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



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From sean at obscuradigital.com  Sat Mar 20 12:21:25 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 20 Mar 2010 12:21:25 -0700
Subject: [Freeswitch-users] Paging
Message-ID: <C7CA6E45.7010%sean@obscuradigital.com>

Hello Dev team,

Freeswitch rocks!!

I have a question about the paging application.  I?ve followed the
instructions on the wiki on how to set it up, but when I do a sip trace I?m
noticing a Proxy authentication message when try to connect to the paging
extension.  Every time I dial the extension it hungs up. I?ve checked the
acl xml file and my subnet is in the access list.  Any thoughts on how to
correct the authentication issue, if that?s issue.

Thanks
Sean
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From mastermind202 at gmail.com  Sat Mar 20 17:01:06 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 20 Mar 2010 20:01:06 -0400
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CA6E45.7010%sean@obscuradigital.com>
References: <C7CA6E45.7010%sean@obscuradigital.com>
Message-ID: <63de75711003201701l49237f2dh153cc4e9ba149018@mail.gmail.com>

On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
> Hello Dev team,
>
> Freeswitch rocks!!
>
> I have a question about the paging application. ?I?ve followed the
> instructions on the wiki on how to set it up, but when I do a sip trace I?m
> noticing a Proxy authentication message when try to connect to the paging
> extension. ?Every time I dial the extension it hungs up. I?ve checked the
> acl xml file and my subnet is in the access list. ?Any thoughts on how to
> correct the authentication issue, if that?s issue.
>
> Thanks
> Sean
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

Sean,

We need more information to help you.  Please include the relevant
piece(s) of your dialplan, the paging hardware, and what version of FS
you are using.  (if you are not using trunk, please update to it to
see if that may fix your problem)

-- mm_202.


From sean at obscuradigital.com  Sat Mar 20 17:35:10 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 20 Mar 2010 17:35:10 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <63de75711003201701l49237f2dh153cc4e9ba149018@mail.gmail.com>
Message-ID: <C7CAB7CE.7017%sean@obscuradigital.com>

Hey mm_202,

Thanks for the response.  Sorry for the lack of info, just getting use to
using freeswitch and not sure what's necessary to troubleshoot all FS
issues.

Here's my dialplan entry:
 <extension name="rtp_multicast_page">
     <condition field="destination_number" expression="^pagegroup$|^7243">
       <action application="answer"/>
       <action application="esf_page_group"/>
     </condition>
   </extension>

When you say "paging hardware" are you talking about my phones? Polycom 321

I just installed FS this morning so I'm sure I have the most recent version.
FYI I'm using freepbx to configure FS.

Do I need to add a dial group to get the paging working? Still learning, but
so far everything else is working.

Sean


On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:

> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello Dev team,
>> 
>> Freeswitch rocks!!
>> 
>> I have a question about the paging application. ?I?ve followed the
>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>> noticing a Proxy authentication message when try to connect to the paging
>> extension. ?Every time I dial the extension it hungs up. I?ve checked the
>> acl xml file and my subnet is in the access list. ?Any thoughts on how to
>> correct the authentication issue, if that?s issue.
>> 
>> Thanks
>> Sean
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> Sean,
> 
> We need more information to help you.  Please include the relevant
> piece(s) of your dialplan, the paging hardware, and what version of FS
> you are using.  (if you are not using trunk, please update to it to
> see if that may fix your problem)
> 
> -- mm_202.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sat Mar 20 17:41:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 19:41:12 -0500
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CAB7CE.7017%sean@obscuradigital.com>
References: <C7CAB7CE.7017%sean@obscuradigital.com>
Message-ID: <5CFB0E7E-191C-494E-8D82-BEAA80496DF1@freeswitch.org>

esf_page_group is multicast paging the polycom doesn't support it.. please see the mad boss examples in the dialplan to perform it for the polycom and just about any other phone.

/b

On Mar 20, 2010, at 7:35 PM, Sean Holt wrote:

> Hey mm_202,
> 
> Thanks for the response.  Sorry for the lack of info, just getting use to
> using freeswitch and not sure what's necessary to troubleshoot all FS
> issues.
> 
> Here's my dialplan entry:
> <extension name="rtp_multicast_page">
>     <condition field="destination_number" expression="^pagegroup$|^7243">
>       <action application="answer"/>
>       <action application="esf_page_group"/>
>     </condition>
>   </extension>
> 
> When you say "paging hardware" are you talking about my phones? Polycom 321
> 
> I just installed FS this morning so I'm sure I have the most recent version.
> FYI I'm using freepbx to configure FS.
> 
> Do I need to add a dial group to get the paging working? Still learning, but
> so far everything else is working.
> 
> Sean
> 
> 
> On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:
> 
>> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>>> Hello Dev team,
>>> 
>>> Freeswitch rocks!!
>>> 
>>> I have a question about the paging application.  I?ve followed the
>>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>>> noticing a Proxy authentication message when try to connect to the paging
>>> extension.  Every time I dial the extension it hungs up. I?ve checked the
>>> acl xml file and my subnet is in the access list.  Any thoughts on how to
>>> correct the authentication issue, if that?s issue.
>>> 
>>> Thanks
>>> Sean
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> Sean,
>> 
>> We need more information to help you.  Please include the relevant
>> piece(s) of your dialplan, the paging hardware, and what version of FS
>> you are using.  (if you are not using trunk, please update to it to
>> see if that may fix your problem)
>> 
>> -- mm_202.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at microcomaustralia.com.au  Sat Mar 20 17:53:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Mar 2010 11:53:30 +1100
Subject: [Freeswitch-users] freeswitch and android
Message-ID: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>

Hello,

Anybody here got freeswitch working with Android?

If so, what app did you use?

I tried sipdroid which looked very good, but unfortunately it seems it
answers incoming phone calls immediately and fakes a ring tone. I
filled a bug report, but it was closed as invalid, so far without
explanation:

http://code.google.com/p/sipdroid/issues/detail?id=383

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>


From brian at freeswitch.org  Sat Mar 20 18:56:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 20:56:03 -0500
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
Message-ID: <7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>

http://files.freeswitch.org/windows_installer/sipdroid.apk

It has SILK support.

/b

On Mar 20, 2010, at 7:53 PM, Brian May wrote:

> Hello,
> 
> Anybody here got freeswitch working with Android?
> 
> If so, what app did you use?
> 
> I tried sipdroid which looked very good, but unfortunately it seems it
> answers incoming phone calls immediately and fakes a ring tone. I
> filled a bug report, but it was closed as invalid, so far without
> explanation:
> 
> http://code.google.com/p/sipdroid/issues/detail?id=383
> 
> Thanks
> -- 
> Brian May <brian at microcomaustralia.com.au>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Sat Mar 20 19:35:28 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Mar 2010 10:35:28 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
Message-ID: <4BA585F0.4050400@coppice.org>

On 03/21/2010 12:02 AM, Brian West wrote:
> I haven't see many of the SOC projects ever produce anything useful as of late.
>    
True, but since they keep choosing the same mentoring organisations each 
year, I assume never producing anything useful is a key part of Google's 
game plan with SoC.

Steve



From yehavi.bourvine at gmail.com  Sat Mar 20 22:54:46 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 21 Mar 2010 07:54:46 +0200
Subject: [Freeswitch-users] How to upgrade configuration files when
	upgrading FreeSwitch
Message-ID: <d361ca7b1003202254i4df16131x9b5cac6e10851750@mail.gmail.com>

Hello,

  Once a week I upgrade our production system by retreiving
freeswitch-1.0.5-latest.tar.gz, building and installing it. This updates all
the binaries, but I am left with the current config files (which is good, as
some of them contains locally edited data). The problem is how to detect
(and merge) changes in the configuration files, like new parameters. I can
do manual diffs, but it is a time consuming task.

What is the BCP of doing such an upgrade?

                      Thanks! __Yehavi:
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From oseslija at gmail.com  Sun Mar 21 02:47:22 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Mar 2010 10:47:22 +0100
Subject: [Freeswitch-users] openzap no audio?
In-Reply-To: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
References: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
Message-ID: <4468a6771003210247r6e74246cqcd0d7e6341f5d86@mail.gmail.com>

I had the same issues with Dahdi. When I was using latest Zaptel, PRI-PRI
calls had audio.

Ognjen

On Thu, Mar 18, 2010 at 6:07 AM, Tom Christensen <pavera at gmail.com> wrote:

> I've got freeswitch installed with a digium te110p hooked up to a PRI.
>
> calls through the system work fine SIP->SIP, calling and receiving calls
> over the PRI works, (IE the signalling part, a call can be made and incoming
> calls come in over the PRI and I can route them based on DID), however,
> there is no audio to either party.  I've checked, and there is an RTP stream
> running while the call is up, there is no reason it would be blocked (IE,
> SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
> few posts that talked about no audio for a few seconds after a call is
> connected, or that calls would connect and then drop after a few seconds.
>  This is different, I can leave the call up for many minutes, it doesn't
> drop, and audio never starts coming through.
>
> I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
> the latest release of zaptel first, freeswitch wouldn't even load with that.
>
> I'm going to try libpri now I think as I've exhausted all the other options
> I can think of.
>
> Any input/ideas would be greatly appreciated.
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ben at langfeld.co.uk  Sun Mar 21 07:42:29 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 21 Mar 2010 14:42:29 +0000
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
	<191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
Message-ID: <fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>

Anthony,

Apologies for the delay in responding here - i've only just got around to
testing this more. After finding a 404 response to BYE, I appear to have
managed to fix this by changing the port the Line1 component of the ATA runs
on to something other than 5060. Looks like everything is working now.

Regards,
Ben Langfeld

Wave > Email


On Fri, Feb 26, 2010 at 8:01 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you would have to be more specific.
> We would need to have you test this on latest SVN trunk with a full debug
> log.
>
> enter the following at your cli and reproduce the call saving all the
> output.
>
> console loglevel debug
> sofia profile internal siptrace on
>
>
>
> On Fri, Feb 26, 2010 at 12:41 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Hey,
>>
>> I have a small 7 seat mostly softphone based PBX installation in place
>> using freeswitch and a couple of SPA3000s for PSTN termination. Currently,
>> aside from PSTN HUP detection, outbound calls are fine setup to directly
>> dial the SPAs. When the internal VoIP side hangs up, freeswitch sends
>> NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely.
>>
>> Unfortunately inbound calls aren't so successful. Calls are sent using the
>> SPA dialplan to an internal freeswitch extension, and the calls connect
>> fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
>> the ATA. Is there a reason for this?
>>
>> Can anyone give me any idea how I can get freeswitch to instruct the SPA
>> to drop the line?
>>
>> Regards,
>> Ben Langfeld
>>
>> Wave > Email
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Sun Mar 21 07:59:20 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Sun, 21 Mar 2010 17:59:20 +0300
Subject: [Freeswitch-users] not registered external softphone's with
	internal profile
Message-ID: <eb0385731003210759j5698f140g88061cf3c66ffc84@mail.gmail.com>

Hello,

I'm trying to register on the fs external softphone (with public ip).
Copied profine internal to internal5090
(http://pastebin.freeswitch.org/12506). When you register get 405
Method Not Allowed or 481 Subscription Does Not Exist
(http://pastebin.freeswitch.org/12507 and
http://pastebin.freeswitch.org/12508). Depending on what softphone
use. sofia_contact shows that the username is not registered
(error/user_not_registered), although it is in sofia status
internal5090 (http://pastebin.freeswitch.org/12509). What causes the
appearance of 405, 481 and the absence of registration?

-- 
Best regards, Vladimir Elizarov


From sean at obscuradigital.com  Sun Mar 21 09:35:14 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sun, 21 Mar 2010 09:35:14 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <5CFB0E7E-191C-494E-8D82-BEAA80496DF1@freeswitch.org>
Message-ID: <C7CB98D2.702F%sean@obscuradigital.com>

Thanks Brian, I'll check out mad_boss and see if it works for my setup.

Also I wanted to point in the wiki documentation for valet_park there's a
potential typo.

 <condition field="destination_number" expression="$(6000)^">

My basic understanding is the $ sign should be at the end not in the front?
I couldn't get this to work until I switched the ^ and $ around.

Sean


On 3/20/10 5:41 PM, "Brian West" <brian at freeswitch.org> wrote:

> esf_page_group is multicast paging the polycom doesn't support it.. please see
> the mad boss examples in the dialplan to perform it for the polycom and just
> about any other phone.
> 
> /b
> 
> On Mar 20, 2010, at 7:35 PM, Sean Holt wrote:
> 
>> Hey mm_202,
>> 
>> Thanks for the response.  Sorry for the lack of info, just getting use to
>> using freeswitch and not sure what's necessary to troubleshoot all FS
>> issues.
>> 
>> Here's my dialplan entry:
>> <extension name="rtp_multicast_page">
>>     <condition field="destination_number" expression="^pagegroup$|^7243">
>>       <action application="answer"/>
>>       <action application="esf_page_group"/>
>>     </condition>
>>   </extension>
>> 
>> When you say "paging hardware" are you talking about my phones? Polycom 321
>> 
>> I just installed FS this morning so I'm sure I have the most recent version.
>> FYI I'm using freepbx to configure FS.
>> 
>> Do I need to add a dial group to get the paging working? Still learning, but
>> so far everything else is working.
>> 
>> Sean
>> 
>> 
>> On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:
>> 
>>> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>>>> Hello Dev team,
>>>> 
>>>> Freeswitch rocks!!
>>>> 
>>>> I have a question about the paging application.  I?ve followed the
>>>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>>>> noticing a Proxy authentication message when try to connect to the paging
>>>> extension.  Every time I dial the extension it hungs up. I?ve checked the
>>>> acl xml file and my subnet is in the access list.  Any thoughts on how to
>>>> correct the authentication issue, if that?s issue.
>>>> 
>>>> Thanks
>>>> Sean
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> Sean,
>>> 
>>> We need more information to help you.  Please include the relevant
>>> piece(s) of your dialplan, the paging hardware, and what version of FS
>>> you are using.  (if you are not using trunk, please update to it to
>>> see if that may fix your problem)
>>> 
>>> -- mm_202.
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Mar 21 09:56:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 11:56:34 -0500
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CB98D2.702F%sean@obscuradigital.com>
References: <C7CB98D2.702F%sean@obscuradigital.com>
Message-ID: <FE269726-9DC4-41BC-85B4-6763AED888C5@freeswitch.org>

Its a wiki.. correct it yourself if you have a moment.

/b

On Mar 21, 2010, at 11:35 AM, Sean Holt wrote:

> Also I wanted to point in the wiki documentation for valet_park there's a
> potential typo.



From brian at freeswitch.org  Sun Mar 21 10:07:39 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 12:07:39 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA585F0.4050400@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
	<4BA585F0.4050400@coppice.org>
Message-ID: <8D3D830E-319A-4A0D-9CC2-772B3ACC1B00@freeswitch.org>

We have applied twice now and didn't get picked either time.

/b

On Mar 20, 2010, at 9:35 PM, Steve Underwood wrote:

> True, but since they keep choosing the same mentoring organisations each 
> year, I assume never producing anything useful is a key part of Google's 
> game plan with SoC.
> 
> Steve



From sean at obscuradigital.com  Sun Mar 21 10:10:40 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sun, 21 Mar 2010 10:10:40 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <FE269726-9DC4-41BC-85B4-6763AED888C5@freeswitch.org>
Message-ID: <C7CBA120.7032%sean@obscuradigital.com>

Didn't know I had that kind of access, but I will gladly.

Sean


On 3/21/10 9:56 AM, "Brian West" <brian at freeswitch.org> wrote:

> Its a wiki.. correct it yourself if you have a moment.
> 
> /b
> 
> On Mar 21, 2010, at 11:35 AM, Sean Holt wrote:
> 
>> Also I wanted to point in the wiki documentation for valet_park there's a
>> potential typo.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Mar 21 10:24:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 12:24:52 -0500
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
	<191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
	<fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>
Message-ID: <91A721BA-A79D-4715-AD76-DD8616AE8DC4@freeswitch.org>

Sounds like you had a SIP ALG in the way.

/b

On Mar 21, 2010, at 9:42 AM, Ben Langfeld wrote:

> Anthony,
> 
> Apologies for the delay in responding here - i've only just got around to testing this more. After finding a 404 response to BYE, I appear to have managed to fix this by changing the port the Line1 component of the ATA runs on to something other than 5060. Looks like everything is working now.
> 
> Regards,
> Ben Langfeld
> 
> Wave > Email



From raffaele.p.guidi at gmail.com  Sun Mar 21 11:35:31 2010
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 21 Mar 2010 19:35:31 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<4BA4D354.3080009@coppice.org>
Message-ID: <d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>

Uhm, I've seen this package (/sip/communicator/impl/media/codec/audio/g729*)
dated back in 2009-06-27 *

*
http://dev.sip-communicator.org/stats/dir_src_net_java_sip_communicator_impl_media_codec_audio_g729.html
*
*
*
*with some 10 thousand lines of code and thought it was there but I'm never
tried that. I also second Brian thoughts about sip comm guys trying to do
too much. But the thing kinda works and even they have not reached any
stable build the phone is usable. Also it is in java - and this helps in
enterprises where C is a bit forgotten. Another good one in this sense was
Sipek 2 (in C#) based on pjsip (I've done real business with that, a sip
enabled browser for a small call center) but too bad development is idle
since last year.*

On Sat, Mar 20, 2010 at 2:53 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> > I believe sip communicator does handle g729. It has a strong community
> > and a plugin model (based on osgi) that makes it easily (ok, kinda...)
> > customizable. Of course the FSComm/givesomemoneytotheproject option is
> > the best one. In any case:
> > http://sip-communicator.org/
> >
> > Ciao,
> >     R
> It seems one of their summer of code projects is to implement a G.729
> codec, so it doesn't seem like they have one.
> >
> > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
> > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
> >
> >     We're looking for an open source softphones that can handle g729
> >     that we can extend with extra functionality (connecting to other
> >     applications, and things on the intranet.)
> >
> >     Does anyone have any they'd suggest?
> >
> >     Brian West suggested FSComm with the qualifier "it still needs
> >     more work" - has anyone used it much (especially in windows), or
> >     know what it's development status is? (Looking in the source
> >     browser jmesquita spent a lot of time on it in January, but not
> >     much since.)
> >
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Mar 21 12:09:33 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 15:09:33 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>

last trunk,

remove current conf folder and reinstalled the default one after make install.
I stopped iptables, connect 1000 to 1001 on port 5060, but
after rtp-timeout it closes yet.

sip trace after the rtp timeout
SIP/2.0 481 Call Does Not Exist


I start to be very frustrated....
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From freeswitch-users-list at metik.com  Sun Mar 21 12:46:43 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Sun, 21 Mar 2010 15:46:43 -0400
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>
References: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>
Message-ID: <4BA677A3.4050407@metik.com>

I haven't followed this particular thread very closely but are you sure 
there are no port conflicts exhibited in the SDP?  This is UDP after all 
and unlike TCP, binding is exclusive and port selection would then need 
to take in account those ports already in use (the endpoint and FS would 
be pretty much flying blind if the endpoint is not a local process).

I would suggest reviewing your endpoint configurations and making sure 
that they are using unique (rtp/media) port ranges.  One would think 
that if the endpoint shares the same global IP address as used by FS and 
it is not a (softphone) process running on the same physical box, then 
it would be communicating with FS using its private IP. 

Does both the signaling and media traffic traverse (i.e. routed) through 
a broadband router that sits in front of both FS and the endpoint or is 
it sent (i.e. forwarded) directly?

-metik

Madovsky wrote:
> last trunk,
>  
> remove current conf folder and reinstalled the default one after make 
> install.
> I stopped iptables, connect 1000 to 1001 on port 5060, but
> after rtp-timeout it closes yet.
>  
> sip trace after the rtp timeout
> SIP/2.0 481 Call Does Not Exist
>  
> I start to be very frustrated....
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From infos at madovsky.org  Sun Mar 21 12:51:04 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 15:51:04 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <4228F02C0FB14F8684B8F057B2BC07FC@MOBILEE1705>

Apparently there s a timeout everytime
I register a phone on a public IP.
for phone behind a firewall it works well.

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 21, 2010 3:09 PM
  Subject: rtp timeout yet


  last trunk,

  remove current conf folder and reinstalled the default one after make install.
  I stopped iptables, connect 1000 to 1001 on port 5060, but
  after rtp-timeout it closes yet.

  sip trace after the rtp timeout
  SIP/2.0 481 Call Does Not Exist


  I start to be very frustrated....
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From justin at ejtown.org  Fri Mar 19 12:17:09 2010
From: justin at ejtown.org (Justin B Newman)
Date: Fri, 19 Mar 2010 15:17:09 -0400
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <00a701cac776$bf147de0$3d3d79a0$@com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
Message-ID: <27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>

> larger customer deployments. If we want to create a trunk to an ITSP and a
> trunk to an internal PBX, would we use the external gateway XML files for
> both? We don?t plan to point any internal phones to the FreeSwitch box
> (we?re using sipX as an internal PBX).

The "internal" and "external" monikers are arbitrary names given to
two SIP profiles.  It's completely possible to run a FreeSWITCH box
with a single profile, named internal, external, or general, or
whatever.  The samples are provided and so-labelled because one of the
biggest problems new users have with PBXs is distinguishing inbound
and outbound calls.  If an inbound call somehow gets into an outbound
dialplan, you can end up with a situation where someone starts using
your PBX to make calls, without your consent.

FreeSWITCH, by permitting association of ports and/or IP addresses
with specific SIP profiles makes it trivial to keep these calls
completely segregated, making it a little harder to do "stupid"
things.

-jbn


From shaikbashaatc at yahoo.com  Sat Mar 20 01:18:09 2010
From: shaikbashaatc at yahoo.com (Shaik basha)
Date: Sat, 20 Mar 2010 01:18:09 -0700 (PDT)
Subject: [Freeswitch-users] Downloading Free Switch
Message-ID: <473732.22231.qm@web43406.mail.sp1.yahoo.com>

Dear All,

Good Afternoon every one in this forum. I am new to FS. I tried to understand about FS, but could not understand completely. Hence would any one help me from where I can download FS and what FS can do exactly. 

Your early response in this regard would be very much appreciated. Thanking you in advance. Regards,

shaikh.bashaa




      



From sashker at sashker.ru  Sat Mar 20 21:31:12 2010
From: sashker at sashker.ru (=?KOI8-R?B?4czFy9PBzsTSIPzE1cHSxM/Xyd4=?=)
Date: Sun, 21 Mar 2010 09:31:12 +0500
Subject: [Freeswitch-users] I need help about my gateway
Message-ID: <aa322b471003202131y6c526f30l9f6d1d9b84652942@mail.gmail.com>

Hello!

I tried to search information about my problem at the internet and local
sites/forums, but I didn't find solution.

My provider need in CONTACT field: "7455 at providerserver.com" (7455 - is
login on provider server), but FS send:
gw+gwname at mydomain.com<gw%2Bgwname at mydomain.com>
".
If I use option "<param name="extension-in-contact" value="true"/>" in my
gateway config - FS send this: "7455 at mydomain.com".

As a result, my external calls didn't send over this provider:

"2010-03-20 17:56:30.852222 [NOTICE] switch_channel.c:669 New Channel
sofia/external/7451 [0ab674ec-bc8c-48e0-9235-
eb395dfe02b0]
2010-03-20 17:56:30.915402 [NOTICE] sofia.c:4788 Hangup sofia/external/7451
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1179 Session 28
(sofia/external/7451) Ended
2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/external/7451 [CS_DESTROY]
2010-03-20 17:56:30.919604 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: NO_ROUTE_DESTINATION
2010-03-20 17:56:30.919604 [NOTICE] mod_dptools.c:2416 Hangup
sofia/internal/1000 at mydomain.com [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1179 Session 27
(sofia/internal/1000 at mydomain.com) Ended
2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/internal/1000 at mydomain.com [CS_DESTROY]"

My gateway config:

"<include>
  <gateway name="gwname">
     <param name="username" value="7455"/>
      <param name="realm" value="providerserver.com"/>
     <param name="from-user" value="7455"/>
      <param name="from-domain" value="providerserver.com"/>
      <param name="password" value="pass"/>
      <param name="extension" value="7455"/>
      <param name="proxy" value="providerserver.com"/>
      <param name="register-proxy" value="providerserver.com"/>
       <param name="expire-seconds" value="160"/>
 <param name="register" value="true"/>
     <param name="register-transport" value="udp"/>
     <param name="retry-seconds" value="30"/>
    <param name="extension-in-contact" value="true"/>
      <param name="caller-id-in-from" value="true"/>
      <param name="contact-params" value="tport=tcp"/>
      <param name="ping" value="25"/>
  </gateway>
</include>

This gateway status in "sofia status": REGED.

Sorry for my bad english )))

Best regards, Alexander
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From emcho at sip-communicator.org  Sun Mar 21 03:05:46 2010
From: emcho at sip-communicator.org (Emil Ivov)
Date: Sun, 21 Mar 2010 11:05:46 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
Message-ID: <4BA5EF7A.9010200@sip-communicator.org>

> sip communicator is crap. Its trying to do way too much and does not
> a single thing very well at all.  

Well, that's a bit vague. Maybe if you've had/have specific issues with
it, you could drop us a note so that we could fix them.

In any case the project has improved a lot during the last years, and
even months so you might want to give it another try one of these days.

> If they have G729 its not legal to use.

Yes we do have it, and it was implemented as a part of our participation
in GSoC 2009. People can use it perfectly legally as long as they buy
the corresponding licenses (just as with any other client).

Currently use of the codec is disabled by default and whoever needs it
has to enable it in the code, after acquiring the necessary license.

Cheers,
Emil

> 
> /b
> 
> On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:
> 
>> I believe sip communicator does handle g729. It has a strong
>> community and a plugin model (based on osgi) that makes it easily
>> (ok, kinda...) customizable. Of course the
>> FSComm/givesomemoneytotheproject option is the best one. In any
>> case: http://sip-communicator.org/
>> 
>> Ciao, R
> 



From edpimentl at gmail.com  Sun Mar 21 16:41:12 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 21 Mar 2010 19:41:12 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA5EF7A.9010200@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
Message-ID: <9dc4a1671003211641q3cc1a088w928cbccea4c1443b@mail.gmail.com>

Here are two softphone supporting g729
http://www.flaphone.com/help.php#about
http://trac.qutecom.org/wiki/SupportedCodecs

Also the above softphone  may provide good UI ideas for FSComm
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From steveu at coppice.org  Sun Mar 21 16:43:43 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 22 Mar 2010 07:43:43 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>
Message-ID: <4BA6AF2F.9090909@coppice.org>

On 03/22/2010 02:35 AM, Raffaele P. Guidi wrote:
> Uhm, I've seen this package 
> (/sip/communicator/impl/media/codec/audio/g729*) dated back 
> in 2009-06-27* **
>
> **http://dev.sip-communicator.org/stats/dir_src_net_java_sip_communicator_impl_media_codec_audio_g729.html
> **
> **
> **
> **with some 10 thousand lines of code and thought it was there but I'm 
> never tried that. I also second Brian thoughts about sip comm guys 
> trying to do too much. But the thing kinda works and even they have 
> not reached any stable build the phone is usable. Also it is in java - 
> and this helps in enterprises where C is a bit forgotten. Another good 
> one in this sense was Sipek 2 (in C#) based on pjsip (I've done real 
> business with that, a sip enabled browser for a small call center) but 
> too bad development is idle since last year.**
That does look like the complete ITU reference codec translated into 
Java. Maybe the SoC project is to move it forwrad, rather than start it. 
The SoC description certainly looks like its to start it. I believe its 
really hard to lock down and licence modules of Java code. I wonder if 
the even have a plan for how this could be legally used.

> On Sat, Mar 20, 2010 at 2:53 PM, Steve Underwood <steveu at coppice.org 
> <mailto:steveu at coppice.org>> wrote:
>
>     On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
>     > I believe sip communicator does handle g729. It has a strong
>     community
>     > and a plugin model (based on osgi) that makes it easily (ok,
>     kinda...)
>     > customizable. Of course the FSComm/givesomemoneytotheproject
>     option is
>     > the best one. In any case:
>     > http://sip-communicator.org/
>     >
>     > Ciao,
>     >     R
>     It seems one of their summer of code projects is to implement a G.729
>     codec, so it doesn't seem like they have one.
>     >
>     > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
>     > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>
>     <mailto:fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>>>
>     wrote:
>     >
>     >     We're looking for an open source softphones that can handle g729
>     >     that we can extend with extra functionality (connecting to other
>     >     applications, and things on the intranet.)
>     >
>     >     Does anyone have any they'd suggest?
>     >
>     >     Brian West suggested FSComm with the qualifier "it still needs
>     >     more work" - has anyone used it much (especially in windows), or
>     >     know what it's development status is? (Looking in the source
>     >     browser jmesquita spent a lot of time on it in January, but not
>     >     much since.)
>     >
>
Steve



From brian at freeswitch.org  Sun Mar 21 16:57:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 18:57:15 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA5EF7A.9010200@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
Message-ID: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>


On Mar 21, 2010, at 5:05 AM, Emil Ivov wrote:

>> sip communicator is crap. Its trying to do way too much and does not
>> a single thing very well at all.  
> 
> Well, that's a bit vague. Maybe if you've had/have specific issues with
> it, you could drop us a note so that we could fix them.

Overall UI behavior is not well tested.  Too many options, too many clicks to actually get to what you want to do.  Video with H264 doesn't work so well from SIP-Communicator to SIP-Communicator as recent as last weeks build.  I'm pretty sure my expectations are higher then most people's as I find bugs in everything I get my hands on.  I usually give up at around two or three bugs.  I understand you're open source as are we and thats hard work.  I'm fairly critical of everything so you can ignore me most of the time.  

> In any case the project has improved a lot during the last years, and
> even months so you might want to give it another try one of these days.

See comment above, Not sure what was going on or what was wrong... it just didn't work as expected or like EyeBeam or X-lite does.

>> If they have G729 its not legal to use.
> 
> Yes we do have it, and it was implemented as a part of our participation
> in GSoC 2009. People can use it perfectly legally as long as they buy
> the corresponding licenses (just as with any other client).

You're facilitating infringement.  Nobody can buy a license legally for a hand full of channels. 

> Currently use of the codec is disabled by default and whoever needs it
> has to enable it in the code, after acquiring the necessary license.

Like they really go get a license, where are they going to do this for one channel?

> Cheers,
> Emil



From brian at microcomaustralia.com.au  Sun Mar 21 17:10:50 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Mar 2010 11:10:50 +1100
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
	<7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>
Message-ID: <3c5cf5261003211710m2692352bh86f584c067d288d@mail.gmail.com>

On 21 March 2010 12:56, Brian West <brian at freeswitch.org> wrote:
> http://files.freeswitch.org/windows_installer/sipdroid.apk

sipdroid seems to be the application I am having problems with.

Unless this is a different version?

When I get a chance I will try and get a packet trace of the problem,
as the only response is that my report is invalid.
-- 
Brian May <brian at microcomaustralia.com.au>


From steveu at coppice.org  Sun Mar 21 17:23:40 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 22 Mar 2010 08:23:40 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
Message-ID: <4BA6B88C.5010201@coppice.org>

On 03/22/2010 07:57 AM, Brian West wrote:
> On Mar 21, 2010, at 5:05 AM, Emil Ivov wrote:
>
>    
>>> sip communicator is crap. Its trying to do way too much and does not
>>> a single thing very well at all.
>>>        
>> Well, that's a bit vague. Maybe if you've had/have specific issues with
>> it, you could drop us a note so that we could fix them.
>>      
> Overall UI behavior is not well tested.  Too many options, too many clicks to actually get to what you want to do.  Video with H264 doesn't work so well from SIP-Communicator to SIP-Communicator as recent as last weeks build.  I'm pretty sure my expectations are higher then most people's as I find bugs in everything I get my hands on.  I usually give up at around two or three bugs.  I understand you're open source as are we and thats hard work.  I'm fairly critical of everything so you can ignore me most of the time.
>
>    
>> In any case the project has improved a lot during the last years, and
>> even months so you might want to give it another try one of these days.
>>      
> See comment above, Not sure what was going on or what was wrong... it just didn't work as expected or like EyeBeam or X-lite does.
>
>    
>>> If they have G729 its not legal to use.
>>>        
>> Yes we do have it, and it was implemented as a part of our participation
>> in GSoC 2009. People can use it perfectly legally as long as they buy
>> the corresponding licenses (just as with any other client).
>>      
> You're facilitating infringement.  Nobody can buy a license legally for a hand full of channels.
>    
Sadly, that's true. For a while you could get one off licences for the 
G.729 patent pool, though I'm not clear if the people outside the pool 
ever offered one off licences for their patents - one isn't much use 
without the other. These days, neither the patents in the pool, or the 
patents outside, can be licenced for small numbers of channels.

I know these patent pools need to be very wary of things like RICO 
charges in the US, but once a standard has become a gating point - 
G.729, H.264, V.34, V.90 etc - we really are better off with 
comprehensive patent pools, overseen by those who police corruption.
>> Currently use of the codec is disabled by default and whoever needs it
>> has to enable it in the code, after acquiring the necessary license.
>>      
> Like they really go get a license, where are they going to do this for one channel?
>
>    
Steve



From emcho at sip-communicator.org  Sun Mar 21 17:47:10 2010
From: emcho at sip-communicator.org (Emil Ivov)
Date: Mon, 22 Mar 2010 01:47:10 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
Message-ID: <4BA6BE0E.1000005@sip-communicator.org>

?? 22.03.10 00:57, Brian West ??????:
>> Well, that's a bit vague. Maybe if you've had/have specific issues
>> with it, you could drop us a note so that we could fix them.
> 
> Overall UI behavior is not well tested.  Too many options, too many
> clicks to actually get to what you want to do. 

Well, I am still not sure I see which options you have in mind so if you
ever come across them again, don't hesitate to drop us (or me) a note.

> Video with H264
> doesn't work so well from SIP-Communicator to SIP-Communicator as
> recent as last weeks build. 

Right, depending on the OS and the device, this one does have some
issues. We have developers looking at them, and expect to see them
resolved in the following couple of weeks. Could let you know when done,
if interested.

> See comment above, Not sure what was going on or what was wrong... it
> just didn't work as expected or like EyeBeam or X-lite does.

Quality-wise, I certainly don't agree. As for look and feel, the
projects are hard to compare since they are simply different
applications with pretty different objectives. This doesn't mean either
of them is ... crap.

Cheers,
Emil



From brian at freeswitch.org  Sun Mar 21 17:49:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 19:49:52 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA6BE0E.1000005@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
	<4BA6BE0E.1000005@sip-communicator.org>
Message-ID: <CF6C2BE1-A17A-4851-A21D-50C3CD5F9D09@freeswitch.org>

Well in general the current state of Soft Phones that just work and look great is lacking.   

/b

On Mar 21, 2010, at 7:47 PM, Emil Ivov wrote:

>> See comment above, Not sure what was going on or what was wrong... it
>> just didn't work as expected or like EyeBeam or X-lite does.
> 
> Quality-wise, I certainly don't agree. As for look and feel, the
> projects are hard to compare since they are simply different
> applications with pretty different objectives. This doesn't mean either
> of them is ... crap.
> 
> Cheers,
> Emil



From devel at thom.fr.eu.org  Mon Mar  1 00:10:24 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Mon, 01 Mar 2010 09:10:24 +0100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
 specifying fxs/fxo-channels
In-Reply-To: <3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
	<3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
Message-ID: <cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>

On Sun, 28 Feb 2010 12:11:33 +1100, Brian May
<brian at microcomaustralia.com.au> wrote:
> On 24 February 2010 20:25, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
>> I use sangoma card and the openzap file is generated by the Setup
script
>> from sangoma driver.
>> It seems that the terminology used by zaptel is not used in wanpipe
>> configuration.
> 
> Yes, that is correct.
> 
>> I have an A400 card with an FXO module (providing ports 11 and 12) and
an
>> FXS module (providing ports 9 and 10)
>>
>> My openzap.conf is like this :
>>
>> [span wanpipe FXS]
>> name => Analog phone 1
>> number => 9000
>> fxs-channel => 1:9
>> name => Analog phone 2
>> number => 9001
>> fxs-channel => 1:10
>>
>> [span wanpipe FXO]
>> name => POTS line 1
>> number => 1234567890
>> fxo-channel => 1:11
>> name => POTS line 2
>> number => 0987654321
>> fxo-channel => 1:12
> 
> So ports 9 and 10 are actually FXO ports - extension ports; ports 11
> and 12 are FXS ports, or telephone lines. This is what I have been
> saying.
> 
> Oh, wait, no it isn't. Looks like I was confused. :-(
> 
> It matches my config however.
> 
> Hopefully this fixed the problems with the wiki:
> 
>
http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18693&oldid=18491

On that point, that looks correct (maybe it would make it clearer with a
sentence between parenthesis sayin "using FXO/FXS signaling").

> 
> My guess is that this change is needed also (not absolutely sure here):
> 

On this one, I would have to check in the code. I don't know if the span
order here makes a difference (my guess is no).

>
http://wiki.freeswitch.org/index.php?title=Openzap.conf_Examples&diff=18694&oldid=18693



From lukasz at voiceworks.pl  Mon Mar  1 01:32:03 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 01 Mar 2010 10:32:03 +0100
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
	<7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
Message-ID: <4B8B8993.7010002@voiceworks.pl>

First disable display IE in smg_pri.conf

disable_display_ie=yes

then try to set TON and NP (you can check what telco is sending in 
incoming SETUP) for example:

callingparty_dialplan=national
calledparty_dialplan=national

Your configuration for telco port should be more like;

signalling=pri_cpe
switchtype=euroisdn
callingparty_dialplan=national
calledparty_dialplan=national
disable_display_ie=yes
group=1
spans=1

of course caller_id_number should be set (also check if you telco 
require 04439114600 or 4439114600).

Regards,
Lukasz

lakshmanan ganapathy wrote:
> Dear Moy,
> That's didn't seem to solve the problem.
> I gave the following command.
> originate 
> {origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266 
> &bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)
>
> The D-Chan Log is
> http://pastebin.freeswitch.org/12268
>
> Kindly refer the attached pcap file that I captured with wanpipemon 
> utility. I think it might help.
>
>
> On Sun, Feb 28, 2010 at 12:05 AM, Moises Silva <moises.silva at gmail.com 
> <mailto:moises.silva at gmail.com>> wrote:
>
>     I believe the problem FreeSWITCH is setting that as a default
>     callerid name, which your telco does not like.
>
>     Try setting the caller id name and number by yourself as explained
>     in the "originate" section
>     here http://wiki.freeswitch.org/wiki/Mod_commands
>
>
>     On Sat, Feb 27, 2010 at 12:13 AM, lakshmanan ganapathy
>     <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>         I think it says Invalid Information Element for the DISPLAY
>         smg_prid/a/8122133885!!!
>         correct?? If so, can you please help me to solve this?
>
>
>         On Sat, Feb 27, 2010 at 10:32 AM, lakshmanan ganapathy
>         <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>             In the Dchan log it is saying Invalid Information
>             Elements. That might be a problem??? But I even don't know
>             why it is saying Invalid Information Element??
>             Please guide me!!!
>
>
>
>             On Sat, Feb 27, 2010 at 10:27 AM, lakshmanan ganapathy
>             <lakindia89 at gmail.com <mailto:lakindia89 at gmail.com>> wrote:
>
>                 Dear Moy,
>                 Here are the details:
>
>                 FreeSwitch Log:
>                 http://pastebin.freeswitch.org/12256
>
>                  /var/log/sangoma_pri/dchan_<span-no>.log:
>                  http://pastebin.freeswitch.org/12257
>
>                 /var/log/sangoma_mgd.log:
>                 http://pastebin.freeswitch.org/12258
>
>                 smg_pri.conf
>                 http://pastebin.freeswitch.org/12259
>
>
>
>                 On Fri, Feb 26, 2010 at 9:01 PM, Moises Silva
>                 <moises.silva at gmail.com
>                 <mailto:moises.silva at gmail.com>> wrote:
>
>                     Hello lakshmanan,
>
>                     Please enable the dchan_log=q931 in
>                     /etc/wanpipe/smg_pri.conf and then restart it
>                     (smg_ctrl restart), then pastebin the logs
>
>                      /var/log/sangoma_pri/dchan_<span-no>.log
>                     /var/log/sangoma_mgd.log
>
>                     That will contain the Q931 details (if any). Also
>                     pastebin your smg_pri.conf.
>
>                     Also enable the FreeSWITCH debug logging (see the
>                     FreeSWITCH wiki for details about that) and paste
>                     them too.
>
>                     -- 
>                     Moises Silva
>                     Senior Software Engineer
>                     Sangoma Technologies Inc. | 50 McIntosh Drive,
>                     Suite 120, Markham ON L3R 9T3 Canada
>                     t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>                     <mailto:moy at sangoma.com>
>
>                     On Fri, Feb 26, 2010 at 12:55 AM, lakshmanan
>                     ganapathy <lakindia89 at gmail.com
>                     <mailto:lakindia89 at gmail.com>> wrote:
>
>                         Dear all,
>                         I'm having a A102 Sangoma hardware. I
>                         configured it with freeswitch.
>                         wanrouter status, says both the port as connected.
>                         My smg_prid version is
>
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
>                         ================System restart=============
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Sangoma PRI Protocol Stack Daemon       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Version: 1.54                           =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Date: Feb 15 2010                       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Wanpipe Release: wanpipe-3.5.8.6       =
>                         Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =
>                         Revision:Revision: 15288                =
>                         Feb 26 16:08:14 FMS-FreeSwitch
>                         sangoma_prid:                              
>                         ===========================================
>
>                         My freeswitch version is 16729.
>                         I started freeswitch.
>
>                         oz list
>                         +OK
>                         span: 1 (smg_prid)
>                         type: Sangoma (boost)
>                         chan_count: 60
>                         dialplan: XML
>                         context: default
>                         dial_regex:
>                         fail_dial_regex:
>                         hold_music:
>                         analog_options none
>
>                         I originated a call as
>                         originate openzap/smg_prid/a/9952248266
>                         &park(), which hits my mobile.
>
>                         But when I issued the following command:
>                         originate openzap/smg_prid/a/9952248266
>                         &bridge(openzap/smg_prid/a/8122133885)
>                         It rings my mobile (9952248266) first, but
>                         after that the following error was displayed
>
>                         2010-02-26 16:20:51.736080 [ERR]
>                         switch_ivr_originate.c:2387 Cannot create
>                         outgoing channel of type [openzap] cause:
>                         [NORMAL_CIRCUIT_CONGESTION]
>                         The call got ended in my mobile.
>
>                         Freeswitch log and smg_pri.conf
>                                   http://pastebin.freeswitch.org/12248
>                         openzap.conf:
>                         [span wanpipe smg_prid]
>                         name => smg_prid
>                         trunk_type =>e1
>                         b-channel => 1:1-15
>                         b-channel => 1:17-31
>                         trunk_type =>e1
>                         b-channel => 2:1-15
>                         b-channel => 2:17-31
>
>                         openzap.conf.xml:
>                         <configuration name="openzap.conf"
>                         description="OpenZAP Configuration">
>                           <settings>
>                             <param name="debug" value="0"/>
>                           </settings>
>                           <boost_spans>
>                                 <span id="smg_prid">
>                                 <param name="dialplan" value="XML"/>
>                                 <param name="context" value="default"/>
>                                 </span>
>                           </boost_spans>
>                         </configuration>
>
>                         Please guide me to setup this one!!.
>
>
>                         _______________________________________________
>                         FreeSWITCH-users mailing list
>                         FreeSWITCH-users at lists.freeswitch.org
>                         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                         http://www.freeswitch.org
>
>
>
>                     _______________________________________________
>                     FreeSWITCH-users mailing list
>                     FreeSWITCH-users at lists.freeswitch.org
>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                     http://www.freeswitch.org
>
>
>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     -- 
>     Moises Silva
>     Senior Software Engineer
>     Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham
>     ON L3R 9T3 Canada
>     t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From michal.kalinowski at interia.pl  Mon Mar  1 03:51:55 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Mon, 1 Mar 2010 12:51:55 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
Message-ID: <7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>

Thank's Brian

So at this moment i have only clean configuration in lua without any XML.

I get example from wiki:

hash={["main"]=undef,

   ["name"]="top",

         ["greet_long"]="phrase:demo_ivr_main_menu",

["greet_short"]="phrase:demo_ivr_main_menu_short",

["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",

["exit_sound"]="voicemail/vmgoodbye.wav",

["confirm_macro"]="undef",

 ["confirm_key"]="undef",

    ["confirm_attempts"]="3",

      ["inter_digit_timeout"]="2000",

       ["digit_len"]="1",

               ["timeout"]="10000",
["max_failures"]="3"
}
top = freeswitch.IVRMenu(hash["main"],
hash["name"],
hash["greet_long"],hash["greet_short"],
hash["invalid_sound"],
hash["exit_sound"],
hash["confirm_macro"],
hash["confirm_key"],
hash["confirm_attempts"],
hash["inter_digit_timeout"],
hash["digit_len"],
hash["timeout"],
hash["max_failures"]);

top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
top:execute(session, "top");

But something is wrong freeswitch say
2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
expected 16..16 args, got 13
stack traceback:
        [C]: in function 'IVRMenu'
        /usr/local/freeswitch/scripts/ivr.lua:19: in main chunk

What args FS expect?

BR,
Micha?
W dniu 27 lutego 2010 22:48 u?ytkownik Brian West
<brian at freeswitch.org> napisa?:
> can you point out on the wiki that indicated you are able to dot his? ?You're mixing two concepts incorrectly here.
>
> For example you can't do "set" and exec the file to get the contents.. you can however on linux use "exec" instead of set. ?But your script needs to print it to stdout. ?Your example in your lua script is for the config engine in in lua thats like XML curl... in which case you're not building the full document like you should. ?Read thru the XML CURL docs if you want to do "XML_STRING = mydialplan".
>
> /b
>
>
> On Feb 27, 2010, at 3:10 PM, michal kalinowski wrote:
>
>> Coming back to this case I create in lua some script with XML ivr.
>>
>> #!/usr/local/bin/lua
>>
>> mydialplan = [[
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <include>
>> ?<!-- demo IVR setup -->
>> ?<!-- demo IVR, Main Menu -->
>> ?<menu name="demo_ivr"
>> ? ? ?greet-long="phrase:demo_ivr_main_menu"
>> ? ? ?greet-short="phrase:demo_ivr_main_menu_short"
>> ? ? ?invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>> ? ? ?exit-sound="voicemail/vm-goodbye.wav"
>> ? ? ?confirm-macro=""
>> ? ? ?confirm-key=""
>> ? ? ?tts-engine="flite"
>> ? ? ?tts-voice="rms"
>> ? ? ?confirm-attempts="3"
>> ? ? ?timeout="10000"
>> ? ? ?inter-digit-timeout="2000"
>> ? ? ?max-failures="3"
>> ? ? ?max-timeouts="3"
>> ? ? ?digit-len="4">
>>
>> ? ?<!-- The following are the definitions for the digits the user dials -->
>> ? ?<!-- Digit 1 transfer caller to the public FreeSWITCH conference -->
>> ? ?<entry action="menu-exec-app" digits="1" param="bridge
>> sofia/$${domain}/888 at conference.freeswitch.org"/>
>> ? ?<entry action="menu-exec-app" digits="2" param="transfer 9996 XML
>> default"/> ? ?<!-- FS echo -->
>> ? ?<entry action="menu-exec-app" digits="3" param="transfer 9999 XML
>> default"/> ? ?<!-- MOH -->
>> ? ?<entry action="menu-exec-app" digits="4" param="transfer 9991 XML
>> default"/> ? ?<!-- ClueCon -->
>> ? ?<entry action="menu-exec-app" digits="5" param="transfer 1234*256
>> enum"/> ? ? ? <!-- Screaming monkeys -->
>> ? ?<entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>
>> ? ? ? ? ? ? <!-- demo sub menu -->
>> ? ?<!-- Using a regex in the digits tag lets you define a dial
>> pattern for the caller
>> ? ? ? ? You may define multiple regexes if you need a different
>> pattern for some reason -->
>> ? ?<entry action="menu-exec-app" digits="/^(10[01][0-9])$/"
>> param="transfer $1 XML features"/>
>> ? ?<entry action="menu-top" digits="9"/> ? ? ? ? ?<!-- Repeat this menu -->
>> ?</menu>
>>
>> </include>
>> ]]
>> XML_STRING = mydialplan
>>
>> in dialplan I have context with this ivr
>>
>> ?<context name="ivr">
>> <extension name="ivr">
>> ? ? <condition field="caller_id_number" expression="^[0-9]"/>
>> ? ? <condition field="destination_number" expression="^(\d{11})$">
>> ? ? ? ?<action application="answer"/>
>> ? ? ? ?<action application="ivr" data="demo_ivr"/>
>> ? ? </condition>
>> </extension>
>> ?</context>
>>
>> in ivr.conf i have this
>>
>> <configuration name="ivr.conf" description="IVR menus">
>> ?<menus>
>> ? ? ? ?<X-PRE-PROCESS cmd="set" data="application=lua lua.ivr"/>
>> ?</menus>
>> </configuration>
>>
>> but for some reasons Freeswitch say "2010-02-27 22:27:48.380342 [ERR]
>> mod_dptools.c:1247 Unable to find menu"
>> what I do wrong ?
>>
>>
>> BR,
>> Micha?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From moises.silva at gmail.com  Mon Mar  1 06:55:57 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 1 Mar 2010 09:55:57 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
References: <7d79b3931002252155t47c86968q9e451482926d93ac@mail.gmail.com>
	<c4d05cbe1002260731v29dbc428i1e72d24a220c62eb@mail.gmail.com>
	<7d79b3931002262057g3b325bctcdbcfb833d4aaf55@mail.gmail.com>
	<7d79b3931002262102w79e16ae1nd786600b44dfb758@mail.gmail.com>
	<7d79b3931002262113s49495e7cga59715f41fdbb35d@mail.gmail.com>
	<c4d05cbe1002271035g6112b277hdf305b61233b94ad@mail.gmail.com>
	<7d79b3931002282216w2e2ec844q36d28b3f50423ec4@mail.gmail.com>
Message-ID: <c4d05cbe1003010655v706a970cxf999c26c3a32a7a0@mail.gmail.com>

Hi again,

On Mon, Mar 1, 2010 at 1:16 AM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Dear Moy,
> That's didn't seem to solve the problem.
> I gave the following command.
> originate
> {origination_caller_id_number=04439114600}openzap/smg_prid/a/9952248266
> &bridge({origination_caller_id_number=04439114600}openzap/smg_prid/a/9976975781)
>

The engineer in charge of the PRI daemon will be sending you an e-mail soon
to fix your problem. Let me know if you don't get the e-mail today.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From mmg at transtelco.net  Mon Mar  1 08:09:05 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 11:09:05 -0500
Subject: [Freeswitch-users] High CPU usage on 1.0.5
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555B40@VMBX113.ihostexchange.net>

Dear freeswitch group

I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU usage even if there are only a few calls on the system or no calls at all. We are running Debian with kernel 2.6.26-2-686

Anyone experimenting a similar issue?


Version
freeswitch at internal> version
FreeSWITCH Version 1.0.5-20100225-0400 (16810M)


Thanks in advance




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From sos at sokhapkin.dyndns.org  Mon Mar  1 08:32:32 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 1 Mar 2010 11:32:32 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
Message-ID: <201003011132.32415.sos@sokhapkin.dyndns.org>

How much is "high" and on which CPU?

On Monday 01 March 2010, Manuel Mar?n wrote:
> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at all.
> We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
>
> Thanks in advance
>
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
>
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214





From brian at freeswitch.org  Mon Mar  1 08:37:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 10:37:27 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <201003011132.32415.sos@sokhapkin.dyndns.org>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
Message-ID: <FB95AD29-58EC-4C98-B744-9E7606EC489C@freeswitch.org>

Please type uname -a and give me the contents of /proc/cpuinfo

/b

On Mar 1, 2010, at 10:32 AM, Sergey Okhapkin wrote:

> How much is "high" and on which CPU?




From mmg at transtelco.net  Mon Mar  1 08:41:26 2010
From: mmg at transtelco.net (=?utf-8?B?TWFudWVsIE1hcsOtbg==?=)
Date: Mon, 1 Mar 2010 11:41:26 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <201003011132.32415.sos@sokhapkin.dyndns.org>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>

Hi Sergey

Attached you will find a screenshot of the CPU usage. There were only 22 calls running. Is it normal on release 1.0.5?


processor       : 0
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 0
initial apicid  : 0
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7188.28
clflush size    : 64
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 3
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 6
initial apicid  : 6
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.45
clflush size    : 64
power management:

processor       : 2
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 1
initial apicid  : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.41
clflush size    : 64
power management:

processor       : 3
vendor_id       : GenuineIntel
cpu family      : 15
model           : 4
model name      : Intel(R) Xeon(TM) CPU 3.60GHz
stepping        : 1
cpu MHz         : 3591.375
cache size      : 1024 KB
physical id     : 3
siblings        : 2
core id         : 0
cpu cores       : 1
apicid          : 7
initial apicid  : 7
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 5
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pebs bts pni monitor ds_cpl est tm2 cid cx16 xtpr
bogomips        : 7182.46
clflush size    : 64
power management:







Manuel Mar?n 
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214


-----Original Message-----
From: Sergey Okhapkin [mailto:sos at sokhapkin.dyndns.org] 
Sent: Monday, March 01, 2010 9:33 AM
To: freeswitch-users at lists.freeswitch.org
Cc: Manuel Mar?n
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5

How much is "high" and on which CPU?

On Monday 01 March 2010, Manuel Mar?n wrote:
> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at all.
> We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
>
> Thanks in advance
>
>
> freeswitch at internal> version
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
>
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214


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From anthony.minessale at gmail.com  Mon Mar  1 08:47:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 10:47:58 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
Message-ID: <191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>

we are still wary about 5.3 due to bugs reported in libc.
What number does it say it detected for the gap?


On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:

> OK - I've realised I do get the same warning with CentOS 5.3, it just goes
> past more quickly so I didn't see it.  Maybe it is just the hardware ....
>
>
> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi,
>>
>> I wonder if anyone would be able to advise please:
>>
>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
>> start FreeSWITCH that
>>
>> "Abnormally large timer gap detected"
>> "Do you have your kernel timer set to greater than 1kHz?  You may
>> experience audio problems".
>>
>> I get no such warning if I build on CentOS 5.3, and the test timings it
>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>> to Centos5.4 was a yum update on the 5.3 build.
>>
>> I guess the warning comes from here:
>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>
>> This is all on pretty low spec hardware - a couple of different Dell
>> optiplex p4's I use for testing.
>>
>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>> 5.4 and not worry about the warnings, or if there is something I can do to
>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>> use such crummy hardware?!
>>
>> Many thanks in advance
>> Bruce
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tomb at cachecomm.com  Mon Mar  1 08:42:39 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 09:42:39 -0700
Subject: [Freeswitch-users] NAT issue?
Message-ID: <4B8BEE7F.1030401@cachecomm.com>

Voip Gods,

I am a newbie to freeswitch i have a problem with nat i think . i can 
only get one sip phone to register with freeswitch here is my setup
FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only 
register phone one . when freeswitch loads it says set ext-ip to my 
gateway IP .  i know  it something super stupid  can some one give me a 
boot in the right direction

Cheers
 Tom



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




From brian at freeswitch.org  Mon Mar  1 08:50:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 10:50:36 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<201003011132.32415.sos@sokhapkin.dyndns.org>
	<4502F03F8260234AB94179D6E1BDD0CF3F53555B79@VMBX113.ihostexchange.net>
Message-ID: <A82574D6-9274-4D61-8E14-D5BAD18C8B85@freeswitch.org>

Please output of uname -a?

/b

On Mar 1, 2010, at 10:41 AM, Manuel Mar?n wrote:

> Attached you will find a screenshot of the CPU usage. There were only 22 calls running. Is it normal on release 1.0.5?




From mmg at transtelco.net  Mon Mar  1 08:53:10 2010
From: mmg at transtelco.net (=?utf-8?B?TWFudWVsIE1hcsOtbg==?=)
Date: Mon, 1 Mar 2010 08:53:10 -0800
Subject: [Freeswitch-users] High CPU usage 1.0.5
Message-ID: <20100301085310.B5C9FD1D@resin15.mta.everyone.net>


Brian

Attached you will find the uname -a output

rtpb2:/var/log# uname -a
Linux rtpb2.transtelco.net 2.6.26-2-686 #1 SMP Wed Feb 10 08:59:21 UTC 2010 i686 GNU/Linux



--- brian at freeswitch.org wrote:

From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5
Date: Mon, 1 Mar 2010 10:37:27 -0600

Please type uname -a and give me the contents of /proc/cpuinfo

/b

On Mar 1, 2010, at 10:32 AM, Sergey Okhapkin wrote:

> How much is "high" and on which CPU?


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





From christian.loeschenkohl at xpirio.com  Mon Mar  1 09:08:17 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 01 Mar 2010 18:08:17 +0100
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
Message-ID: <4B8BF481.5000400@xpirio.com>

hello

have you tried the -vm startup flag?

anthony stated on an similar question (from me)
---
If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that may not manifest into the load average.
---

i also use a new trunk version, after using -vm the load where normal again

br


On 2010-03-01 06:20, Manuel Mar?n wrote:

> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at
> all. We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
> Thanks in advance
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
> Manuel Mar?n
>
> Transtelco
>
> US 1.915.2172232
>
> MX 52.656.6921109
>
> FAX 1.915.2311214
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From msc at freeswitch.org  Mon Mar  1 09:24:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 09:24:56 -0800
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8BEE7F.1030401@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
Message-ID: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>

On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com> wrote:

> Voip Gods,
>
> I am a newbie to freeswitch i have a problem with nat i think . i can
> only get one sip phone to register with freeswitch here is my setup
> FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
> register phone one . when freeswitch loads it says set ext-ip to my
> gateway IP .  i know  it something super stupid  can some one give me a
> boot in the right direction
>
> Cheers
>  Tom
>
> What device is performing the NAT? Make sure it doesn't have an ALG in
there as well. See this new page that we have been updating:
http://wiki.freeswitch.org/wiki/ALG

Also, which phones are you using?
-MC
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From mmg at transtelco.net  Mon Mar  1 09:25:08 2010
From: mmg at transtelco.net (=?iso-8859-1?Q?Manuel_Mar=EDn?=)
Date: Mon, 1 Mar 2010 12:25:08 -0500
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4B8BF481.5000400@xpirio.com>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<4B8BF481.5000400@xpirio.com>
Message-ID: <4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>

Thanks Christian. I'll try with the -vm startup flag. I think that issue is related to timing because even if there are no calls running on the system the CPU load is 50-70%




Manuel Mar?n
Transtelco
US 1.915.2172232
MX 52.656.6921109
FAX 1.915.2311214


-----Original Message-----
From: Christian L?schenkohl [mailto:christian.loeschenkohl at xpirio.com]
Sent: Monday, March 01, 2010 10:08 AM
To: freeswitch-users at lists.freeswitch.org
Cc: Manuel Mar?n
Subject: Re: [Freeswitch-users] High CPU usage 1.0.5

hello

have you tried the -vm startup flag?

anthony stated on an similar question (from me)
---
If you have a box that has trouble with timing it could cost more resources.
you can always run freeswitch -vm to use an alternate form of timing that may not manifest into the load average.
---

i also use a new trunk version, after using -vm the load where normal again

br


On 2010-03-01 06:20, Manuel Mar?n wrote:

> Dear freeswitch group
>
> I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> usage even if there are only a few calls on the system or no calls at
> all. We are running Debian with kernel 2.6.26-2-686
>
> Anyone experimenting a similar issue?
>
> Thanks in advance
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
>
> Manuel Mar?n
>
> Transtelco
>
> US 1.915.2172232
>
> MX 52.656.6921109
>
> FAX 1.915.2311214
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


--
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From codecomplete at free.fr  Mon Mar  1 10:08:00 2010
From: codecomplete at free.fr (Fred-145)
Date: Mon, 01 Mar 2010 19:08:00 +0100
Subject: [Freeswitch-users] [init.d script] Possible improvement?
Message-ID: <ve0oo5hcev0ij4hbjkpg8liju4ua7159fp@4ax.com>

Hello

I use the sample Init script in the wiki to have Freeswitch start
automatically at boot-time on my test CentOS 5.4 host, but I notice
that it takes several seconds for FS to be up and running before I can
connect to it with fs_cli:

========
# /etc/init.d/freeswitch restart
Shutting down freeswitch: [  OK  ]
Starting freeswitch: [  OK  ]

# fs_cli
[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket
Connection Error]
========

I don't know much about Init scripts: Is there a way to display some
information so I know when the script is done and FS is ready accept
commands from the CLI?

Thank you.




From anthony.minessale at gmail.com  Mon Mar  1 10:38:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 12:38:59 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002182154o52352c62h8c0ec02fa66e8444@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
Message-ID: <191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>

I am trying my best not to be completely annoyed at the "somewhat limiting"
remark after going over the countless hours I spent making sure this feature
work and then being patient with 2 people insisting there was a bug when
they both shared an identical misconfiguration.

If we started doing DNS lookups everywhere so hostname could be interchanged
with IP we would have large blocking all over the place when the DNS was not
available and horribly messy code doing all the lookups.  There was already
an experiment called project asterisk that proved the flaws in this design
approach.

FreeSWITCH requires that you keep the hostname uniform for all configuration
be it a domain name or an IP so long as its identical everywhere.  an IP
*is* a domain in sip it's not another form of some DNS lookup.


On Sun, Feb 28, 2010 at 12:23 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> This is one of the most important feature my users want. They don't care
> how I do it, they are just happy it works.
>
>                         Thanks! __Yehavi:
>
> 2010/2/28 Brian West <brian at freeswitch.org>
>
> Really?  Come on guys... the feature is something you can't find elsewhere
>> without paying and you're all still not totally pleased with it?
>>
>> /me shakes his head.
>>
>> We have already started talking about how to make the feature more robust.
>>
>> /b
>>
>> On Feb 28, 2010, at 11:56 AM, Yehavi Bourvine wrote:
>>
>> > Hello Gabe,
>> >
>> >   I agree that this is somewhat limiting, but with Polycom's central
>> provisioning (via XML files) I don't see this as a major drawback.
>> >
>> >                          __Yehavi:
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Mar  1 11:09:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 11:09:30 -0800
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
	<7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
Message-ID: <87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>

2010/3/1 michal kalinowski <michal.kalinowski at interia.pl>

> Thank's Brian
>
> So at this moment i have only clean configuration in lua without any XML.
>
> I get example from wiki:
>
> hash={["main"]=undef,
>
>   ["name"]="top",
>
>         ["greet_long"]="phrase:demo_ivr_main_menu",
>
> ["greet_short"]="phrase:demo_ivr_main_menu_short",
>
> ["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",
>
> ["exit_sound"]="voicemail/vmgoodbye.wav",
>
> ["confirm_macro"]="undef",
>
>  ["confirm_key"]="undef",
>
>    ["confirm_attempts"]="3",
>
>      ["inter_digit_timeout"]="2000",
>
>       ["digit_len"]="1",
>
>               ["timeout"]="10000",
> ["max_failures"]="3"
> }
> top = freeswitch.IVRMenu(hash["main"],
> hash["name"],
> hash["greet_long"],hash["greet_short"],
> hash["invalid_sound"],
> hash["exit_sound"],
> hash["confirm_macro"],
> hash["confirm_key"],
> hash["confirm_attempts"],
> hash["inter_digit_timeout"],
> hash["digit_len"],
> hash["timeout"],
> hash["max_failures"]);
>
> top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
> top:execute(session, "top");
>
> But something is wrong freeswitch say
> 2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
> expected 16..16 args, got 13
> stack traceback:
>        [C]: in function 'IVRMenu'
>        /usr/local/freeswitch/scripts/ivr.lua:19: in main chunk
>
> What args FS expect?
>

I ran into this issue when I was converting the demo IVR to Lua. There are
indeed 16 arguments but you've only got 13 of them. Here's a snippet from
the Lua script I'm working on:
ivr_def = {
    ["main"]                = undef,
    ["name"]                = "demo_ivr_lua",
    ["greet_long"]          = "phrase:demo_ivr_main_menu",
    ["greet_short"]         = "phrase:demo_ivr_main_menu_short",
    ["invalid_sound"]       = "ivr/ivr-that_was_an_invalid_entry.wav",
    ["exit_sound"]          = "voicemail/vm-goodbye.wav",
    ["confirm_macro"]       = "",
    ["confirm_key"]         = "",
    ["tts_engine"]          = "flite",
    ["tts_voice"]           = "rms",
    ["confirm_attempts"]    = "3",
    ["inter_digit_timeout"] = "2000",
    ["digit_len"]           = "4",
    ["timeout"]             = "10000",
    ["max_failures"]        = "3",
    ["max_timeouts"]        = "2"
}
Those are all 16 arguments. Notice you need tts_engine, tts_voice, and
max_timeouts. Try that and let us know how it goes.

-MC
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From anthony.minessale at gmail.com  Mon Mar  1 11:29:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:29:37 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
Message-ID: <191c3a031003011129q777468d0ve852e3ad7363dd3a@mail.gmail.com>

Lesson 1,

NAT Issue is a redundant statement like Failure Problem



On Mon, Mar 1, 2010 at 11:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com> wrote:
>
>> Voip Gods,
>>
>> I am a newbie to freeswitch i have a problem with nat i think . i can
>> only get one sip phone to register with freeswitch here is my setup
>> FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
>> register phone one . when freeswitch loads it says set ext-ip to my
>> gateway IP .  i know  it something super stupid  can some one give me a
>> boot in the right direction
>>
>> Cheers
>>  Tom
>>
>> What device is performing the NAT? Make sure it doesn't have an ALG in
> there as well. See this new page that we have been updating:
> http://wiki.freeswitch.org/wiki/ALG
>
> Also, which phones are you using?
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Mon Mar  1 11:30:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 1 Mar 2010 21:30:26 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<2F87B95C-EA23-4072-B845-3F7AE0617BE3@jerris.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
Message-ID: <d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>

Hello Anthony,


 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>

> I am trying my best not to be completely annoyed at the "somewhat limiting"
> remark after going over the countless hours I spent making sure this feature
> work and then being patient with 2 people insisting there was a bug when
> they both shared an identical misconfiguration.
>
"somewhat limiting" was because my experiments up to now with Polycom and
SIP/TLS were successfull with domain names only. However, I am going to
inverstigate this further to see how this can work with IP only.

In the meantime I've documented the required configuration under the Polycom
wiki page.


> If we started doing DNS lookups everywhere so hostname could be
> interchanged with IP we would have large blocking all over the place when
> the DNS was not available and horribly messy code doing all the lookups.
> There was already an experiment called project asterisk that proved the
> flaws in this design approach.
>
> I think that's why some of us move from Asterisk to FreeSwitch. I also
preffer less to no DNS lookups on a critical system which must work even
when there are some kinds of communication problems.

                      Thanks, __Yehavi:
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From anthony.minessale at gmail.com  Mon Mar  1 11:31:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:31:46 -0600
Subject: [Freeswitch-users] High CPU usage 1.0.5
In-Reply-To: <4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>
References: <4502F03F8260234AB94179D6E1BDD0CF3F53555A73@VMBX113.ihostexchange.net>
	<4B8BF481.5000400@xpirio.com>
	<4502F03F8260234AB94179D6E1BDD0CF3F53555BBD@VMBX113.ihostexchange.net>
Message-ID: <191c3a031003011131q5ea99122w1b987ee0d94979a3@mail.gmail.com>

update to SVN trunk where that option is now default.

If you are getting a lot of cpu usage on an idle box,
update both FS and upgrade or downgrade your kernel away from the "tickless
bug"


On Mon, Mar 1, 2010 at 11:25 AM, Manuel Mar?n <mmg at transtelco.net> wrote:

> Thanks Christian. I'll try with the -vm startup flag. I think that issue is
> related to timing because even if there are no calls running on the system
> the CPU load is 50-70%
>
>
>
>
> Manuel Mar?n
> Transtelco
> US 1.915.2172232
> MX 52.656.6921109
> FAX 1.915.2311214
>
>
> -----Original Message-----
> From: Christian L?schenkohl [mailto:christian.loeschenkohl at xpirio.com]
> Sent: Monday, March 01, 2010 10:08 AM
> To: freeswitch-users at lists.freeswitch.org
> Cc: Manuel Mar?n
> Subject: Re: [Freeswitch-users] High CPU usage 1.0.5
>
> hello
>
> have you tried the -vm startup flag?
>
> anthony stated on an similar question (from me)
> ---
> If you have a box that has trouble with timing it could cost more
> resources.
> you can always run freeswitch -vm to use an alternate form of timing that
> may not manifest into the load average.
> ---
>
> i also use a new trunk version, after using -vm the load where normal again
>
> br
>
>
> On 2010-03-01 06:20, Manuel Mar?n wrote:
>
> > Dear freeswitch group
> >
> > I Just upgraded a system from 1.0.4 to 1.0.5 and we are seeing high CPU
> > usage even if there are only a few calls on the system or no calls at
> > all. We are running Debian with kernel 2.6.26-2-686
> >
> > Anyone experimenting a similar issue?
> >
> > Thanks in advance
> >
> > freeswitch at internal> version
> >
> > FreeSWITCH Version 1.0.5-20100225-0400 (16810M)
> >
> > Manuel Mar?n
> >
> > Transtelco
> >
> > US 1.915.2172232
> >
> > MX 52.656.6921109
> >
> > FAX 1.915.2311214
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar  1 11:42:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 13:42:39 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002182346k71fd602fh6e6725834a6de528@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
Message-ID: <191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>

you can use names if you use *all names*
you just can't mix, but all IP is much faster and safer because there is no
need to do constant dns lookups on something that never changes anyway, you
can easily alias your profile to a friendly name, it also makes it more
portable.




On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello Anthony,
>
>
>  2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>
>> I am trying my best not to be completely annoyed at the "somewhat
>> limiting" remark after going over the countless hours I spent making sure
>> this feature work and then being patient with 2 people insisting there was a
>> bug when they both shared an identical misconfiguration.
>>
> "somewhat limiting" was because my experiments up to now with Polycom and
> SIP/TLS were successfull with domain names only. However, I am going to
> inverstigate this further to see how this can work with IP only.
>
> In the meantime I've documented the required configuration under the
> Polycom wiki page.
>
>
>> If we started doing DNS lookups everywhere so hostname could be
>> interchanged with IP we would have large blocking all over the place when
>> the DNS was not available and horribly messy code doing all the lookups.
>> There was already an experiment called project asterisk that proved the
>> flaws in this design approach.
>>
>> I think that's why some of us move from Asterisk to FreeSwitch. I also
> preffer less to no DNS lookups on a critical system which must work even
> when there are some kinds of communication problems.
>
>                       Thanks, __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tomb at cachecomm.com  Mon Mar  1 12:44:44 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 13:44:44 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
Message-ID: <4B8C273C.6010808@cachecomm.com>

Michael Collins wrote:
>
>
> On Mon, Mar 1, 2010 at 8:42 AM, Tom <tomb at cachecomm.com 
> <mailto:tomb at cachecomm.com>> wrote:
>
>     Voip Gods,
>
>     I am a newbie to freeswitch i have a problem with nat i think . i can
>     only get one sip phone to register with freeswitch here is my setup
>     FS(public ip) -> internet ->NAT -> phones(priveta ip) . it will only
>     register phone one . when freeswitch loads it says set ext-ip to my
>     gateway IP .  i know  it something super stupid  can some one give
>     me a
>     boot in the right direction
>
>     Cheers
>      Tom
>
> What device is performing the NAT? Make sure it doesn't have an ALG in 
> there as well. See this new page that we have been updating:
> http://wiki.freeswitch.org/wiki/ALG
>
> Also, which phones are you using?
> -MC
>
>
>
MC,
the phone are both Aastra they are behind a netgear fvs338 which from 
your post are very bad.  I will try to disable the ALG if i can and try 
again.

Thanks for the help

Cheers
Tom


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




From brian at freeswitch.org  Mon Mar  1 12:51:22 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 14:51:22 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8C273C.6010808@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
Message-ID: <FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>

The only disable button on a netgear is to a sledge hammer and go buy something that doesn't mess with your SIP traffic  :P

/b

On Mar 1, 2010, at 2:44 PM, Tom wrote:

> MC,
> the phone are both Aastra they are behind a netgear fvs338 which from 
> your post are very bad.  I will try to disable the ALG if i can and try 
> again.
> 
> Thanks for the help
> 
> Cheers
> Tom




From gkuri at ieee.org  Mon Mar  1 13:09:39 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 1 Mar 2010 13:09:39 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<191c3a031002190908r3922efccqc6d223a5fabd458b@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
Message-ID: <8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>

Hi Anthony,

I appreciate all the hard work yourself, Brian, Mike and all the other
developers have put in FS! By no means am I discounting any of it,
it's truly invaluable.

Our "philosophy" has always been to use DNS where available, so that
we're not worrying about re-configuring a bunch of devices if IP
addresses change and this obviously goes to setting the phones up as
well to use DNS for the proxy and registrar fields. And it allows us
to do fancy things, like split-view DNS, where if a phone is taken
off-site and we want it to remotely register to FS, it will get the
public IP address of the FS server, rather than the private, without
re-configuring the phone.

If I'm understanding your email correctly, you're saying we need to
use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
DNS, but perhaps I'm missing something somewhere. How would we be able
to use the DNS name of our SIP domain in phones (rather than IP) and
ensure SLA works for both incoming/outgoing calls? Is that possible at
this point in time or do we need to use the IP addresses in the proxy
field on the phone if we want to use SLA?

Thanks for all your help.

Gabriel Kuri


On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> you can use names if you use *all names*
> you just can't mix, but all IP is much faster and safer because there is no
> need to do constant dns lookups on something that never changes anyway, you
> can easily alias your profile to a friendly name, it also makes it more
> portable.
>
>
>
>
> On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>>
>> Hello Anthony,
>>
>> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>>>
>>> I am trying my best not to be completely annoyed at the "somewhat
>>> limiting" remark after going over the countless hours I spent making sure
>>> this feature work and then being patient with 2 people insisting there was a
>>> bug when they both shared an identical misconfiguration.
>>
>> "somewhat limiting" was because my experiments up to now with Polycom and
>> SIP/TLS were successfull with domain names only. However, I am going to
>> inverstigate this further to see how this can work with IP only.
>>
>> In the meantime I've documented the required configuration under the
>> Polycom wiki page.
>>
>>>
>>> If we started doing DNS lookups everywhere so hostname could be
>>> interchanged with IP we would have large blocking all over the place when
>>> the DNS was not available and horribly messy code doing all the lookups.
>>> There was already an experiment called project asterisk that proved the
>>> flaws in this design approach.
>>>
>> I think that's why some of us move from Asterisk to FreeSwitch. I also
>> preffer less to no DNS lookups on a critical system which must?work even
>> when there are some kinds of communication problems.
>>
>> ????????????????????? Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Mon Mar  1 13:22:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 15:22:55 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<d361ca7b1002280149i18e66cbdo2031e7f385f23075@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
Message-ID: <191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>

I really want to be done with this thread......

You were using the IP in some places and the hostnames in others, this makes
it so the SQL statements not match because sql statements cannot do dns
lookups.  I will not be doing any work to resolve hostnames back and forth
to IP inside FS.


The subscribes were using the hostname and the db and invites were using the
IP.

you have to have the same exact domain in your user directory and have the
force-register-domain and force-register-domain-db also set to that value or
omit them completely and use the domain names in 100% of the config fields
on the phone.




On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri at ieee.org> wrote:

> Hi Anthony,
>
> I appreciate all the hard work yourself, Brian, Mike and all the other
> developers have put in FS! By no means am I discounting any of it,
> it's truly invaluable.
>
> Our "philosophy" has always been to use DNS where available, so that
> we're not worrying about re-configuring a bunch of devices if IP
> addresses change and this obviously goes to setting the phones up as
> well to use DNS for the proxy and registrar fields. And it allows us
> to do fancy things, like split-view DNS, where if a phone is taken
> off-site and we want it to remotely register to FS, it will get the
> public IP address of the FS server, rather than the private, without
> re-configuring the phone.
>
> If I'm understanding your email correctly, you're saying we need to
> use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
> DNS, but perhaps I'm missing something somewhere. How would we be able
> to use the DNS name of our SIP domain in phones (rather than IP) and
> ensure SLA works for both incoming/outgoing calls? Is that possible at
> this point in time or do we need to use the IP addresses in the proxy
> field on the phone if we want to use SLA?
>
> Thanks for all your help.
>
> Gabriel Kuri
>
>
> On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > you can use names if you use *all names*
> > you just can't mix, but all IP is much faster and safer because there is
> no
> > need to do constant dns lookups on something that never changes anyway,
> you
> > can easily alias your profile to a friendly name, it also makes it more
> > portable.
> >
> >
> >
> >
> > On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com>
> > wrote:
> >>
> >> Hello Anthony,
> >>
> >> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
> >>>
> >>> I am trying my best not to be completely annoyed at the "somewhat
> >>> limiting" remark after going over the countless hours I spent making
> sure
> >>> this feature work and then being patient with 2 people insisting there
> was a
> >>> bug when they both shared an identical misconfiguration.
> >>
> >> "somewhat limiting" was because my experiments up to now with Polycom
> and
> >> SIP/TLS were successfull with domain names only. However, I am going to
> >> inverstigate this further to see how this can work with IP only.
> >>
> >> In the meantime I've documented the required configuration under the
> >> Polycom wiki page.
> >>
> >>>
> >>> If we started doing DNS lookups everywhere so hostname could be
> >>> interchanged with IP we would have large blocking all over the place
> when
> >>> the DNS was not available and horribly messy code doing all the
> lookups.
> >>> There was already an experiment called project asterisk that proved the
> >>> flaws in this design approach.
> >>>
> >> I think that's why some of us move from Asterisk to FreeSwitch. I also
> >> preffer less to no DNS lookups on a critical system which must work even
> >> when there are some kinds of communication problems.
> >>
> >>                       Thanks, __Yehavi:
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From wchao at yahoo.com  Mon Mar  1 13:31:29 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Mon, 1 Mar 2010 16:31:29 -0500 (EST)
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
Message-ID: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>

I am using the xml_rpc interface like so:

http://192.168.1.1:8080/webapi/show?calls

I get a response in a table, which is a little bit undesirable because I 
want to parse the result in a program and don't want to parse a bunch of 
HTML (just seems inelegant and wasteful), so I instead tried the following 
request:

http://192.168.1.1:8080/txtapi/show?calls

That works better, but then another wrinkle appears. Caller ID (and Callee 
ID) names sometimes have commas, which messes up the CSV.

This appears to be a bug in mod_commands.c (or I suppose you could call it 
an artifact of somebody coding up something quickly and not worrying about 
the absolute correctness of something that is not critical to the core). 
There is also another bug with show calls via webapi in that the generated 
HTML has spurious <td> tags. I think the spurious <td> tag bug arises due 
to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally, 
another bug (or maybe just unimplemented feature) is that xmlapl isn't 
really different from webapi.

These problems occur in the latest source tree (20100301).

I am wondering if (a) one of the developers can fix these or (b) if I fix 
the bugs, how can I submit patches [and if I submit patches, will they be 
accepted into the main tree]?

The fixes are pretty trivial and I'd be happy to code them up if somebody 
will tell me how I can submit the patches (haven't done it before).



From m.sobkow at marketelsystems.com  Mon Mar  1 13:34:15 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 01 Mar 2010 15:34:15 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
Message-ID: <4B8C32D7.40202@marketelsystems.com>

I'm using the originate command from Erlang, specifying a timeout of 30 
seconds.  It is properly giving up on dialing, but it's still returning 
a call UUID instead of an error/timed out status.

How can I query Freeswitch to determine whether the call was answered or 
timed out, without resorting to the event interface?  (I could put 
together a dialplan that calls the event interface after doing the 
originate, but from what I've seen that event interface won't execute if 
the previous command "fails", such as when I did a play_and_get_digits 
timing out.)

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




From brian at freeswitch.org  Mon Mar  1 13:48:45 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 15:48:45 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <777F5906-C26D-4D22-B238-426CB833B88A@freeswitch.org>

Just off the top of my head have you tried /xmlapi/? or read the wiki page on this?  http://wiki.freeswitch.org/wiki/Mod_xml_rpc

and "show calls as xml"?

/b


On Mar 1, 2010, at 3:31 PM, Wellie Chao wrote:

> I am using the xml_rpc interface like so:
> 
> http://192.168.1.1:8080/webapi/show?calls
> 
> I get a response in a table, which is a little bit undesirable because I 
> want to parse the result in a program and don't want to parse a bunch of 
> HTML (just seems inelegant and wasteful), so I instead tried the following 
> request:
> 
> http://192.168.1.1:8080/txtapi/show?calls
> 
> That works better, but then another wrinkle appears. Caller ID (and Callee 
> ID) names sometimes have commas, which messes up the CSV.
> 
> This appears to be a bug in mod_commands.c (or I suppose you could call it 
> an artifact of somebody coding up something quickly and not worrying about 
> the absolute correctness of something that is not critical to the core). 
> There is also another bug with show calls via webapi in that the generated 
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due 
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally, 
> another bug (or maybe just unimplemented feature) is that xmlapl isn't 
> really different from webapi.
> 
> These problems occur in the latest source tree (20100301).
> 
> I am wondering if (a) one of the developers can fix these or (b) if I fix 
> the bugs, how can I submit patches [and if I submit patches, will they be 
> accepted into the main tree]?
> 
> The fixes are pretty trivial and I'd be happy to code them up if somebody 
> will tell me how I can submit the patches (haven't done it before).

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From msc at freeswitch.org  Mon Mar  1 13:52:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 13:52:41 -0800
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>

On Mon, Mar 1, 2010 at 1:31 PM, Wellie Chao <wchao at yahoo.com> wrote:

> I am using the xml_rpc interface like so:
>
> http://192.168.1.1:8080/webapi/show?calls
>
> I get a response in a table, which is a little bit undesirable because I
> want to parse the result in a program and don't want to parse a bunch of
> HTML (just seems inelegant and wasteful), so I instead tried the following
> request:
>
> http://192.168.1.1:8080/txtapi/show?calls
>
> That works better, but then another wrinkle appears. Caller ID (and Callee
> ID) names sometimes have commas, which messes up the CSV.
>
> This appears to be a bug in mod_commands.c (or I suppose you could call it
> an artifact of somebody coding up something quickly and not worrying about
> the absolute correctness of something that is not critical to the core).
> There is also another bug with show calls via webapi in that the generated
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
> another bug (or maybe just unimplemented feature) is that xmlapl isn't
> really different from webapi.
>
> These problems occur in the latest source tree (20100301).
>
> I am wondering if (a) one of the developers can fix these or (b) if I fix
> the bugs, how can I submit patches [and if I submit patches, will they be
> accepted into the main tree]?
>
> The fixes are pretty trivial and I'd be happy to code them up if somebody
> will tell me how I can submit the patches (haven't done it before).
>

Go to jira.freeswitch.org and sign up for an account. You can open an issue,
assign it to yourself, and then submit a patch. Be sure to upload .txt files
and not tar, gz, or bz2! Also, check out the bug reporting tips on this
page:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

It gives some more details on reporting bugs and the things that you should
be doing.
-MC
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From brian at freeswitch.org  Mon Mar  1 13:56:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 15:56:32 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<87f2f3b91003011352r6ee2189em48926d5fb3e45346@mail.gmail.com>
Message-ID: <6E086D9C-0968-4E86-B954-8247FD6B32E4@freeswitch.org>

He should be doing "show channels as xml" via XML API if he wants to parse them. .. or better yet doing it via real XMLRPC.  ;)

/b

On Mar 1, 2010, at 3:52 PM, Michael Collins wrote:

> Go to jira.freeswitch.org and sign up for an account. You can open an issue, assign it to yourself, and then submit a patch. Be sure to upload .txt files and not tar, gz, or bz2! Also, check out the bug reporting tips on this page:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> It gives some more details on reporting bugs and the things that you should be doing.
> -MC
> 

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From andrew at hijacked.us  Mon Mar  1 14:15:24 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 1 Mar 2010 17:15:24 -0500
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8C32D7.40202@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
Message-ID: <20100301221524.GB1751@hijacked.us>

On Mon, Mar 01, 2010 at 03:34:15PM -0600, Mark Sobkow wrote:
> I'm using the originate command from Erlang, specifying a timeout of 30 
> seconds.  It is properly giving up on dialing, but it's still returning 
> a call UUID instead of an error/timed out status.
> 
How are you doing an originate, and what 'UUID' are you getting back?

The way I do it is I do an bgapi originate with origination_uuid set and
I poll for that UUID 10 times every 100 milliseconds to see if its up.
Even if you originate a channel that's going to ring out the channel
will be up for that period and so you can connect to it and receive the
HANGUP event with the cause code to see what happened.

See the init() function in

http://github.com/Vagabond/OpenACD/blob/master/src/freeswitch_ring.erl

for an example. Its ugly but effective.

Andrew



From msc at freeswitch.org  Mon Mar  1 14:29:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 14:29:18 -0800
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
Message-ID: <87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>

On Sun, Feb 28, 2010 at 3:36 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Is there anyway to detect energy levels on a channel that is being
> controlled by mod_lua? Please point me in the right direction if there is.
> Thanks.
>
> Is there a way to do it on a channel that is not controlled by Lua?!
-MC
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From robert.hadley at teotech.com  Mon Mar  1 14:44:21 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 1 Mar 2010 14:44:21 -0800
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
	FXOline
In-Reply-To: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
References: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
Message-ID: <7524ABB6A6684383A76BB9E8B55C679A@greyhawk.tonecommander.com>

 

Using a Sangoma A200 FXO card, is there any way for Freeswitch to
automatically detect a disconnected CO line and use another FXO channel?

 

Thanks,

Robert

 

  _____  

From: Robert Hadley [mailto:robert.hadley at teotech.com] 
Sent: Thursday, February 25, 2010 4:13 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
FXOline

 

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
channel is disconnected and tries dialing out on the channel anyway.  No
error is reported.  The call doesn't timeout until a minute later.
Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next
connected channel?

 

I have configured a Sangoma A200 FXO card as a FXO span.

 

[span wanpipe FXO]

name => PSTN Line 1

number => 4253491059

fxo-channel => 2:3

name => PSTN Line 2

number => 4253491058

fxo-channel => 2:4

 

 

The wanpipe driver does detect and report when a CO line is connected or
disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
reported in the log.

 

/var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
FXO Line is disconnected!

FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_TRAP][3:1] STATE [DOWN]

 

/var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
FXO Line is connected!

FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
[ALARM_CLEAR][3:2] STATE [DOWN]

 

 

I have the dialplan configured to use the next available port in the FXO
span (there will be more than 2 channels later).

 

<extension name="OutgoingFXO">

  <condition field="destination_number" expression="^7(\d+)$">

    <action application="bridge" data="openzap/FXO/a/$1"/>

  </condition>

</extension>

 

 

Here is a portion of the log when that shows dialing out on a disconnected
analog FXO channel.

 

EXECUTE sofia/internal/5410 at 192.168.72.45:5060
bridge(openzap/FXO/a/93491045)

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
channel OpenZAP/3:1/93491045

2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045)
State Change CS_NEW -> CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
from DOWN to DIALING

2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
OpenZAP/3:1/93491045 does not support the proxy feature, disabling.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
starting.

2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
handler on 3:1 for DIALING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_INIT

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
State Change CS_INIT -> CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/3:1/93491045) State INIT going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING

2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
CHANNEL ROUTING

2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
OpenZAP/3:1/93491045 [BREAK]

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/3:1/93491045) State ROUTING going to sleep

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA

2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep

 

 

Thanks,

Robert

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From mrene_lists at avgs.ca  Mon Mar  1 14:59:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 17:59:31 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
Message-ID: <0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>

An easy way is to do an average of the absolute value of each L16  
sample over a period of time. Im not quite sure how to access audio  
channels from within Lua but if you are in C you can use  
switch_core_session_read_frame() to get audio frames.

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




On 1-Mar-10, at 5:29 PM, Michael Collins wrote:

>
>
> On Sun, Feb 28, 2010 at 3:36 AM, Matthew Fong <mattdfong at gmail.com>  
> wrote:
> Is there anyway to detect energy levels on a channel that is being  
> controlled by mod_lua? Please point me in the right direction if  
> there is. Thanks.
>
> Is there a way to do it on a channel that is not controlled by Lua?!
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Mon Mar  1 15:24:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Mar 2010 17:24:16 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
Message-ID: <191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>

Perhaps you may be reading up on something quickly and not worrying about
the absolute correctness of something that is not critical to our time to
explain.

The proper syntax for the command is "show calls as xml" and you are not
really using XMLRPC you are hitting it via direct url. so you will then
need.

http://192.168.1.1:8080/txtapi/show?calls%20as%20xml<http://192.168.1.1:8080/txtapi/show?calls>


anything in the show command can be replied to with "as xml"

also, for fun, try

show calls as csv :::




On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:

> I am using the xml_rpc interface like so:
>
> http://192.168.1.1:8080/webapi/show?calls
>
> I get a response in a table, which is a little bit undesirable because I
> want to parse the result in a program and don't want to parse a bunch of
> HTML (just seems inelegant and wasteful), so I instead tried the following
> request:
>
> http://192.168.1.1:8080/txtapi/show?calls
>
> That works better, but then another wrinkle appears. Caller ID (and Callee
> ID) names sometimes have commas, which messes up the CSV.
>
> This appears to be a bug in mod_commands.c (or I suppose you could call it
> an artifact of somebody coding up something quickly and not worrying about
> the absolute correctness of something that is not critical to the core).
> There is also another bug with show calls via webapi in that the generated
> HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
> to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
> another bug (or maybe just unimplemented feature) is that xmlapl isn't
> really different from webapi.
>
> These problems occur in the latest source tree (20100301).
>
> I am wondering if (a) one of the developers can fix these or (b) if I fix
> the bugs, how can I submit patches [and if I submit patches, will they be
> accepted into the main tree]?
>
> The fixes are pretty trivial and I'd be happy to code them up if somebody
> will tell me how I can submit the patches (haven't done it before).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tculjaga at gmail.com  Mon Mar  1 15:37:42 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Mar 2010 00:37:42 +0100
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
Message-ID: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>

what about changing the destination ports on FS .. 5080 instead of 5060 ...
and make your phone register on that?

I think it could trick the router.

T.

On Mon, Mar 1, 2010 at 9:51 PM, Brian West <brian at freeswitch.org> wrote:

> The only disable button on a netgear is to a sledge hammer and go buy
> something that doesn't mess with your SIP traffic  :P
>
> /b
>
> On Mar 1, 2010, at 2:44 PM, Tom wrote:
>
> > MC,
> > the phone are both Aastra they are behind a netgear fvs338 which from
> > your post are very bad.  I will try to disable the ALG if i can and try
> > again.
> >
> > Thanks for the help
> >
> > Cheers
> > Tom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  1 15:52:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 17:52:05 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
Message-ID: <77FA7E8F-EACD-4262-A27D-ED11674E82B5@freeswitch.org>

Unless its EVIL and looks at all ports like some of them do.  :P

/b

On Mar 1, 2010, at 5:37 PM, Tihomir Culjaga wrote:

> what about changing the destination ports on FS .. 5080 instead of 5060 ... and make your phone register on that?
> 
> I think it could trick the router.
> 
> T.




From msc at freeswitch.org  Mon Mar  1 16:15:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 16:15:12 -0800
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
Message-ID: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>

On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> An easy way is to do an average of the absolute value of each L16 sample
> over a period of time. Im not quite sure how to access audio channels from
> within Lua but if you are in C you can use switch_core_session_read_frame()
> to get audio frames.
>

Show off! :) Related question: do media bugs allow you to access this kind
of information? Just curious...
-MC
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From mrene_lists at avgs.ca  Mon Mar  1 16:20:47 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 19:20:47 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>
	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>
	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
Message-ID: <8D25A218-B53B-48B2-BBF3-2E6ABA9E161B@avgs.ca>

Media bugs allow you to get a channel's audio so you can use that yeah.

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




On 1-Mar-10, at 7:15 PM, Michael Collins wrote:

>
>
> On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> An easy way is to do an average of the absolute value of each L16  
> sample over a period of time. Im not quite sure how to access audio  
> channels from within Lua but if you are in C you can use  
> switch_core_session_read_frame() to get audio frames.
>
> Show off! :) Related question: do media bugs allow you to access  
> this kind of information? Just curious...
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tomb at cachecomm.com  Mon Mar  1 16:24:14 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 01 Mar 2010 17:24:14 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
Message-ID: <4B8C5AAE.2030908@cachecomm.com>

Tihomir Culjaga wrote:
> what about changing the destination ports on FS .. 5080 instead of 
> 5060 ... and make your phone register on that?
>
> I think it could trick the router.
>
> T.
>
> On Mon, Mar 1, 2010 at 9:51 PM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     The only disable button on a netgear is to a sledge hammer and go
>     buy something that doesn't mess with your SIP traffic  :P
>
>     /b
>
>     On Mar 1, 2010, at 2:44 PM, Tom wrote:
>
>     > MC,
>     > the phone are both Aastra they are behind a netgear fvs338 which
>     from
>     > your post are very bad.  I will try to disable the ALG if i can
>     and try
>     > again.
>     >
>     > Thanks for the help
>     >
>     > Cheers
>     > Tom
>
>
Tihomir,
I have tried 5080, 5090  and 5080 still the same result.


Tom 


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




From brian at freeswitch.org  Mon Mar  1 16:27:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 18:27:14 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8C5AAE.2030908@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
Message-ID: <45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>

Sledge hammer it is.

/b

On Mar 1, 2010, at 6:24 PM, Tom wrote:

> Tihomir,
> I have tried 5080, 5090  and 5080 still the same result.
> 
> 
> Tom 




From peder at networkoblivion.com  Mon Mar  1 16:31:14 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 1 Mar 2010 18:31:14 -0600
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel
	with	mod_lua
In-Reply-To: <87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
Message-ID: <070d01cab99f$aac3ae90$004b0bb0$@com>

What's a "media bug"?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 01, 2010 6:15 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Detecting Energy Levels on a Channel with
mod_lua

 

 

On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

An easy way is to do an average of the absolute value of each L16 sample
over a period of time. Im not quite sure how to access audio channels from
within Lua but if you are in C you can use switch_core_session_read_frame()
to get audio frames.


Show off! :) Related question: do media bugs allow you to access this kind
of information? Just curious...
-MC

 

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From mrene_lists at avgs.ca  Mon Mar  1 16:35:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 1 Mar 2010 19:35:31 -0500
Subject: [Freeswitch-users] Detecting Energy Levels on a Channel with
	mod_lua
In-Reply-To: <070d01cab99f$aac3ae90$004b0bb0$@com>
References: <4256bf831002280336vb7268e2oe48b0f74bac335f8@mail.gmail.com>	<87f2f3b91003011429i6b228bddm176e26045c724cab@mail.gmail.com>	<0FF93093-1543-4331-AB2A-BFC8BA5CFA55@avgs.ca>
	<87f2f3b91003011615v305783cfw154e177c590d6e9@mail.gmail.com>
	<070d01cab99f$aac3ae90$004b0bb0$@com>
Message-ID: <D6E7975C-55FD-496A-A2A6-6221B13D11D3@avgs.ca>

A media bug is a way to get a channel's audio inside a module, you can  
have read, write and read_replace/write_replace bugs (those can  
actually change the audio).

Its what record_session uses to be able to record the call to a file  
while its executing other applications.
Other examples are... tone_detect, start_dtmf, eavesdrop...

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




On 1-Mar-10, at 7:31 PM, Peder wrote:

> What?s a ?media bug??
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Monday, March 01, 2010 6:15 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Detecting Energy Levels on a Channel  
> with mod_lua
>
>
>
> On Mon, Mar 1, 2010 at 2:59 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> An easy way is to do an average of the absolute value of each L16  
> sample over a period of time. Im not quite sure how to access audio  
> channels from within Lua but if you are in C you can use  
> switch_core_session_read_frame() to get audio frames.
>
> Show off! :) Related question: do media bugs allow you to access  
> this kind of information? Just curious...
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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

Hello folks!

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

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

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

-Michael
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From david.yatsin at gmail.com  Mon Mar  1 17:34:12 2010
From: david.yatsin at gmail.com (Davud Yat Sin)
Date: Mon, 1 Mar 2010 20:34:12 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
Message-ID: <4b8c6b16.9753f10a.4c0f.ffff9ce5@mx.google.com>

Hi lakshmanan, 
This was fixed in Openzap revision:1047

Can you update your openzap and update to these wanpipe drivers:
ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.6.smg_pri-v1.63.tgz

 

 

David Yat Sin, BEng, Software Engineer

Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 119 | e. dyatsin at sangoma.com
www.sangoma.com  | wiki.sangoma.com

 

 

Dear all,
I'm having a A102 Sangoma hardware. I configured it with freeswitch.
wanrouter status, says both the port as connected.
My smg_prid version is
 
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System
restart=============
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack
Daemon       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version:
1.54                           =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15
2010                       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release:
wanpipe-3.5.8.6       =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision:
15288                =
Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid:
===========================================
 
My freeswitch version is 16729.
I started freeswitch.
 
oz list
+OK
span: 1 (smg_prid)
type: Sangoma (boost)
chan_count: 60
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none
 
I originated a call as
originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile.
 
But when I issued the following command:
originate openzap/smg_prid/a/9952248266
&bridge(openzap/smg_prid/a/8122133885)
It rings my mobile (9952248266) first, but after that the following error
was displayed
 
2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
The call got ended in my mobile.
 
Freeswitch log and smg_pri.conf
          http://pastebin.freeswitch.org/12248
openzap.conf:
[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31
 
openzap.conf.xml:
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
  </boost_spans>
</configuration>
 
Please guide me to setup this one!!.

 

 

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From pjintheusa at gmail.com  Mon Mar  1 18:35:28 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Mar 2010 21:35:28 -0500
Subject: [Freeswitch-users] .NET / ESL
Message-ID: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>

Hi there,

I am sure that I have read somewhere that someone is developing a new .NET
library for FreeSWITCH ESL. If so can someone point me to some info / code
etc. so I can take a look.

Thanks

pj
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From brian at freeswitch.org  Mon Mar  1 18:42:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Mar 2010 20:42:41 -0600
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
References: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
Message-ID: <81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>

http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/

in the managed folder.

/b

On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:

> Hi there,
> 
> I am sure that I have read somewhere that someone is developing a new .NET library for FreeSWITCH ESL. If so can someone point me to some info / code etc. so I can take a look.
> 
> Thanks
> 
> pj
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From pjintheusa at gmail.com  Mon Mar  1 19:01:48 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 1 Mar 2010 22:01:48 -0500
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>
References: <367751821003011835n248e353ayab8536d59f7491ff@mail.gmail.com>
	<81D995F7-5575-497D-A064-6251C4B18774@freeswitch.org>
Message-ID: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>

Thanks

On Mon, Mar 1, 2010 at 9:42 PM, Brian West <brian at freeswitch.org> wrote:

> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
>
> in the managed folder.
>
> /b
>
> On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
>
> > Hi there,
> >
> > I am sure that I have read somewhere that someone is developing a new
> .NET library for FreeSWITCH ESL. If so can someone point me to some info /
> code etc. so I can take a look.
> >
> > Thanks
> >
> > pj
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From delorenzodesign at gmail.com  Mon Mar  1 19:17:50 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Mon, 1 Mar 2010 22:17:50 -0500
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback doesn't seem to get called
Message-ID: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>

I've got the following Lua script working in a sense, but the InputCallback
never seems to get called while the file is being streamed to the call
recipient.  I've tried moving the "vmd start" command and set input callback
around a bit, but to no avail.  I'm testing this against a cell phone voice
mailbox (Verizon).

freeswitch.consoleLog("info","########################################################\n\n");
>
> number_to_call = argv[1]
> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>
> voicemail_detected = false;
>
> function onInput(s, type, obj)
>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
> *************\n");
>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
> session:getVariable("vmd_detect") .. " *************\n");
>
>     if(type == "event" and voicemail_detected == false) then
>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
> MACHINE DETECTED *************\n");
>         voicemail_detected = true;
>         return "break";
>     end
> end
>
> function playbackMessage()
>     sleep_time = 1000;
>     if(voicemail_detected) then
>        sleep_time = 2500;
>     end
>     -- sleep a second
>     session:sleep(sleep_time);
>     -- play a file
>     session:streamFile(message_to_play);
> end
>
> function notify()
>     session =
> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
> .. number_to_call)
>
>
>
>     if(session:ready()) then
>         -- answer the call
>         session:answer();
>         session:setInputCallback("onInput", "");
>         session:execute("vmd","start");
>
>         playbackMessage();
>         if(voicemail_detected) then
>             freeswitch.consoleLog("notice","************ DOING PLAYBACK FOR
> VOICEMAIL/ANSWERING MACHINE *************\n");
>             playbackMessage();
>         end
>
>         freeswitch.consoleLog("notice", "********* hanging up session
> **********\n");
>         -- hangup
>         session:hangup();
>     end
> end
>
> notify();
>
>
> freeswitch.consoleLog("info","########################################################\n\n");
>

-- 
Michael De Lorenzo
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From wchao at yahoo.com  Mon Mar  1 19:29:24 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Mon, 1 Mar 2010 22:29:24 -0500 (EST)
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>

Terrific. "show calls as xml" works great and just what I needed.

P.S. "show calls as csv" still has the comma bug. Not a big deal since I 
can get it as XML. CSV needs quotes around field values that contain a 
comma. Pretty easy to fix. Few lines of C (strchr for "," and if found, 
put quotes around field value).


Date: Mon, 1 Mar 2010 17:24:16 -0600
From: Anthony Minessale <anthony.minessale at gmail.com>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bug in mod_commands.c with show calls command

Perhaps you may be reading up on something quickly and not worrying about
the absolute correctness of something that is not critical to our time to explain.

The proper syntax for the command is "show calls as xml" and you are not really using XMLRPC you are hitting it via direct url.
so you will then need.

http://192.168.1.1:8080/txtapi/show?calls%20as%20xml


anything in the show command can be replied to with "as xml"

also, for fun, try

show calls as csv :::




On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
      I am using the xml_rpc interface like so:

      http://192.168.1.1:8080/webapi/show?calls

      I get a response in a table, which is a little bit undesirable because I
      want to parse the result in a program and don't want to parse a bunch of
      HTML (just seems inelegant and wasteful), so I instead tried the following
      request:

      http://192.168.1.1:8080/txtapi/show?calls

      That works better, but then another wrinkle appears. Caller ID (and Callee
      ID) names sometimes have commas, which messes up the CSV.

      This appears to be a bug in mod_commands.c (or I suppose you could call it
      an artifact of somebody coding up something quickly and not worrying about
      the absolute correctness of something that is not critical to the core).
      There is also another bug with show calls via webapi in that the generated
      HTML has spurious <td> tags. I think the spurious <td> tag bug arises due
      to lines 3145-6 in mod/applications/mod_commands/mod_commands.c. Finally,
      another bug (or maybe just unimplemented feature) is that xmlapl isn't
      really different from webapi.

      These problems occur in the latest source tree (20100301).

      I am wondering if (a) one of the developers can fix these or (b) if I fix
      the bugs, how can I submit patches [and if I submit patches, will they be
      accepted into the main tree]?

      The fixes are pretty trivial and I'd be happy to code them up if somebody
      will tell me how I can submit the patches (haven't done it before).

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




--
Anthony Minessale II

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

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

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




From infos at madovsky.org  Mon Mar  1 20:38:40 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 1 Mar 2010 23:38:40 -0500
Subject: [Freeswitch-users] server to server jingle
Message-ID: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>

Hi all,

is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
I read this article
http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
but no example
and this one
http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
but seems to doesn't work as expected.

Thanks !

Franck
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From msc at freeswitch.org  Mon Mar  1 21:07:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Mar 2010 21:07:22 -0800
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback 	doesn't seem to get called
In-Reply-To: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
Message-ID: <87f2f3b91003012107w59b21e6m2a86100866bb14d3@mail.gmail.com>

I haven't played with events inside Lua yet, but here's an example:
http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts

Look at the onInput() function in that example. There it gives the syntax
for extracting the event header to see what event type has come in.

The other question I have: did you confirm that mod_vmd triggers properly
when going to this mailbox? You could write a simple dialplan that sets
mod_vmd, calls the phone, sleeps for 10 or so seconds, and then dumps the
var ${vmd_detect} to see what it says. I've used mod_vmd from the XML
dialplan with a fair amount of success, although to my knowledge I never
actually tried it with a Verizon wireless VM box.

Confirm that mod_vmd works with just the dialplan before trying to debug the
Lua script.

-MC

On Mon, Mar 1, 2010 at 7:17 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I've got the following Lua script working in a sense, but the InputCallback
> never seems to get called while the file is being streamed to the call
> recipient.  I've tried moving the "vmd start" command and set input callback
> around a bit, but to no avail.  I'm testing this against a cell phone voice
> mailbox (Verizon).
>
>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>>  number_to_call = argv[1]
>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>
>> voicemail_detected = false;
>>
>> function onInput(s, type, obj)
>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>> *************\n");
>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>> session:getVariable("vmd_detect") .. " *************\n");
>>
>>      if(type == "event" and voicemail_detected == false) then
>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>> MACHINE DETECTED *************\n");
>>         voicemail_detected = true;
>>         return "break";
>>     end
>> end
>>
>> function playbackMessage()
>>     sleep_time = 1000;
>>     if(voicemail_detected) then
>>        sleep_time = 2500;
>>     end
>>     -- sleep a second
>>     session:sleep(sleep_time);
>>     -- play a file
>>     session:streamFile(message_to_play);
>> end
>>
>> function notify()
>>     session =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>> .. number_to_call)
>>
>>
>>
>>     if(session:ready()) then
>>         -- answer the call
>>         session:answer();
>>         session:setInputCallback("onInput", "");
>>         session:execute("vmd","start");
>>
>>         playbackMessage();
>>         if(voicemail_detected) then
>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>             playbackMessage();
>>         end
>>
>>         freeswitch.consoleLog("notice", "********* hanging up session
>> **********\n");
>>         -- hangup
>>         session:hangup();
>>     end
>> end
>>
>> notify();
>>
>>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>
> --
> Michael De Lorenzo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gkuri at ieee.org  Mon Mar  1 21:49:06 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 1 Mar 2010 21:49:06 -0800
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
	<191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
Message-ID: <8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>

I've torn my hair out for hours and I still can't get it to work with
the domain name set everywhere <sigh> ...

I can get it to work if I hardcode the IP address on the phones and in
all the config files, but I can't get the domain name to work on
incoming calls, I'm at a total loss. If I had a mismatch somewhere,
I'd think it wouldn't work on outgoing calls either, but that seems to
work fine.

Any other ideas? Might it have anything to do with the incoming call
coming in from the PSTN and going through the external profile?

Cheers,
Gabe

On Mon, Mar 1, 2010 at 1:22 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I really want to be done with this thread......
>
> You were using the IP in some places and the hostnames in others, this makes
> it so the SQL statements not match because sql statements cannot do dns
> lookups.? I will not be doing any work to resolve hostnames back and forth
> to IP inside FS.
>
>
> The subscribes were using the hostname and the db and invites were using the
> IP.
>
> you have to have the same exact domain in your user directory and have the
> force-register-domain and force-register-domain-db also set to that value or
> omit them completely and use the domain names in 100% of the config fields
> on the phone.
>
>
>
>
> On Mon, Mar 1, 2010 at 3:09 PM, Gabriel Kuri <gkuri at ieee.org> wrote:
>>
>> Hi Anthony,
>>
>> I appreciate all the hard work yourself, Brian, Mike and all the other
>> developers have put in FS! By no means am I discounting any of it,
>> it's truly invaluable.
>>
>> Our "philosophy" has always been to use DNS where available, so that
>> we're not worrying about re-configuring a bunch of devices if IP
>> addresses change and this obviously goes to setting the phones up as
>> well to use DNS for the proxy and registrar fields. And it allows us
>> to do fancy things, like split-view DNS, where if a phone is taken
>> off-site and we want it to remotely register to FS, it will get the
>> public IP address of the FS server, rather than the private, without
>> re-configuring the phone.
>>
>> If I'm understanding your email correctly, you're saying we need to
>> use all IP or all DNS for SLA, is that correct? AFAIK, we're using all
>> DNS, but perhaps I'm missing something somewhere. How would we be able
>> to use the DNS name of our SIP domain in phones (rather than IP) and
>> ensure SLA works for both incoming/outgoing calls? Is that possible at
>> this point in time or do we need to use the IP addresses in the proxy
>> field on the phone if we want to use SLA?
>>
>> Thanks for all your help.
>>
>> Gabriel Kuri
>>
>>
>> On Mon, Mar 1, 2010 at 11:42 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > you can use names if you use *all names*
>> > you just can't mix, but all IP is much faster and safer because there is
>> > no
>> > need to do constant dns lookups on something that never changes anyway,
>> > you
>> > can easily alias your profile to a friendly name, it also makes it more
>> > portable.
>> >
>> >
>> >
>> >
>> > On Mon, Mar 1, 2010 at 1:30 PM, Yehavi Bourvine
>> > <yehavi.bourvine at gmail.com>
>> > wrote:
>> >>
>> >> Hello Anthony,
>> >>
>> >> 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>
>> >>>
>> >>> I am trying my best not to be completely annoyed at the "somewhat
>> >>> limiting" remark after going over the countless hours I spent making
>> >>> sure
>> >>> this feature work and then being patient with 2 people insisting there
>> >>> was a
>> >>> bug when they both shared an identical misconfiguration.
>> >>
>> >> "somewhat limiting" was because my experiments up to now with Polycom
>> >> and
>> >> SIP/TLS were successfull with domain names only. However, I am going to
>> >> inverstigate this further to see how this can work with IP only.
>> >>
>> >> In the meantime I've documented the required configuration under the
>> >> Polycom wiki page.
>> >>
>> >>>
>> >>> If we started doing DNS lookups everywhere so hostname could be
>> >>> interchanged with IP we would have large blocking all over the place
>> >>> when
>> >>> the DNS was not available and horribly messy code doing all the
>> >>> lookups.
>> >>> There was already an experiment called project asterisk that proved
>> >>> the
>> >>> flaws in this design approach.
>> >>>
>> >> I think that's why some of us move from Asterisk to FreeSwitch. I also
>> >> preffer less to no DNS lookups on a critical system which must?work
>> >> even
>> >> when there are some kinds of communication problems.
>> >>
>> >> ????????????????????? Thanks, __Yehavi:
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Mon Mar  1 22:01:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 00:01:38 -0600
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support
In-Reply-To: <8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>
References: <87f2f3b91001111742k691966cfs82515db77c13d4d4@mail.gmail.com>
	<8b1c9cda1002280947qcb5ec5anfeef645b447b1866@mail.gmail.com>
	<d361ca7b1002280956g8ef520bw5f4fb891cfc2d473@mail.gmail.com>
	<987F87B1-FA49-47B4-9823-5A7140459AFD@freeswitch.org>
	<d361ca7b1002281023s38d81815t48d985aa830a87c9@mail.gmail.com>
	<191c3a031003011038y233bf1a5p62e00c0474f1d617@mail.gmail.com>
	<d361ca7b1003011130l3c86f240x39ec71462a447318@mail.gmail.com>
	<191c3a031003011142y296e5f14i7e6165123200b92d@mail.gmail.com>
	<8b1c9cda1003011309q598c88d7kb89d7f901f2551ca@mail.gmail.com>
	<191c3a031003011322i54e57ae6n1c8d9a9dc55a42b8@mail.gmail.com>
	<8b1c9cda1003012149u2783538du272d6ce86b9e8f14@mail.gmail.com>
Message-ID: <3DE15C61-E4E3-4EC2-A319-20350092C8F0@freeswitch.org>

You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen.

Verify the subscribe packet...

/b

On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:

> I've torn my hair out for hours and I still can't get it to work with
> the domain name set everywhere <sigh> ...
> 
> I can get it to work if I hardcode the IP address on the phones and in
> all the config files, but I can't get the domain name to work on
> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
> I'd think it wouldn't work on outgoing calls either, but that seems to
> work fine.
> 
> Any other ideas? Might it have anything to do with the incoming call
> coming in from the PSTN and going through the external profile?
> 
> Cheers,
> Gabe




From infos at madovsky.org  Mon Mar  1 22:05:17 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 01:05:17 -0500
Subject: [Freeswitch-users] wiki broken link pocketsphinx
Message-ID: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>

Hello,

the link that goes to grammar sphinx is broken.
the right link is

http://cmusphinx.sourceforge.net/sphinx4/

Regards

Franck
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From infos at madovsky.org  Mon Mar  1 22:24:35 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 01:24:35 -0500
Subject: [Freeswitch-users] last svn version compile error
Message-ID: <5887BE899D4F47B0BF05FCB3340B58CB@MOBILEE1705>

Hi,

Ino succes to update freeswitch to version 16859
with usual command "make current" etc...
but now I got this error :

/usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of ?switch_core_media_bug_add? makes integer from pointer without a cast
/usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function ?switch_core_media_bug_add?

Any idea ?

thanks

Franck

Linux Fedora 10  64bits
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From dujinfang at gmail.com  Mon Mar  1 22:57:51 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Mar 2010 14:57:51 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
Message-ID: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>

I have iLBC configured in profile internal

CODECS IN               PCMU,PCMA,iLBC at 30i,GSM
CODECS OUT              PCMU,PCMA,iLBC at 30i,GSM

and     <param name="inbound-codec-negotiation" value="generous"/>

but it seems not comparing with iLBC

 010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[PCMU:0:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[PCMA:8:8000:20]
2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:30]/[GSM:3:8000:20]


can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .

Thanks.



From dujinfang at gmail.com  Mon Mar  1 23:01:28 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Mar 2010 15:01:28 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
In-Reply-To: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
Message-ID: <23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>

sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)


2010/3/2 Seven Du <dujinfang at gmail.com>:
> I have iLBC configured in profile internal
>
> CODECS IN ? ? ? ? ? ? ? PCMU,PCMA,iLBC at 30i,GSM
> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,iLBC at 30i,GSM
>
> and ? ? <param name="inbound-codec-negotiation" value="generous"/>
>
> but it seems not comparing with iLBC
>
> ?010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
> payload to 101
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>
>
> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>
> Thanks.
>



From lakindia89 at gmail.com  Mon Mar  1 23:14:28 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 2 Mar 2010 12:44:28 +0530
Subject: [Freeswitch-users] make current - System Hangs
Message-ID: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>

Hi all,
I did make current from my freeswitch src directory, but the system got
hanged.
Any one know why this happened.

Here is the output that is printed when I run make current
http://pastebin.freeswitch.org/12279
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From tculjaga at gmail.com  Tue Mar  2 01:03:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Mar 2010 10:03:27 +0100
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
Message-ID: <65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>

bastard !!!! :)

T.

On Tue, Mar 2, 2010 at 1:27 AM, Brian West <brian at freeswitch.org> wrote:

> Sledge hammer it is.
>
> /b
>
> On Mar 1, 2010, at 6:24 PM, Tom wrote:
>
> > Tihomir,
> > I have tried 5080, 5090  and 5080 still the same result.
> >
> >
> > Tom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sharad at coraltele.com  Tue Mar  2 03:20:31 2010
From: sharad at coraltele.com (Sharad)
Date: Tue, 2 Mar 2010 03:20:31 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch & T.38
Message-ID: <1267528831989-4660389.post@n2.nabble.com>


Hi friends,

Is Freeswitch 1.05 carrying T.38 support also ? 

If not, do we have any community member who can provide paid T.38 support
with Freeswitch ?


-- 
View this message in context: http://n2.nabble.com/Freeswitch-T-38-tp4660389p4660389.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From michal.kalinowski at interia.pl  Tue Mar  2 03:28:47 2010
From: michal.kalinowski at interia.pl (michal kalinowski)
Date: Tue, 2 Mar 2010 12:28:47 +0100
Subject: [Freeswitch-users] ivr from mysql
In-Reply-To: <87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>
References: <7c74f5761002150824i41a2ed5did008f36cbb4499a4@mail.gmail.com>
	<87f2f3b91002151002v6bbc3e7fgb67dd6d3e72a915@mail.gmail.com>
	<7c74f5761002151425g512fa685n13c58d8f4b894740@mail.gmail.com>
	<0FD30AAA-C2E2-4228-A09B-DB09F4E26811@avgs.ca>
	<7c74f5761002151525p7cc690ces632d46c6ab5e236c@mail.gmail.com>
	<E9EBF591-5DEC-4FF3-B2D8-20DDA54B4FF5@jerris.com>
	<7c74f5761002271310h7e925f7cy25dd69ba60b896ef@mail.gmail.com>
	<23CC9D8A-65D5-438F-B117-00FEC087418D@freeswitch.org>
	<7c74f5761003010351u4f3d41cbua3bbf4909c00b551@mail.gmail.com>
	<87f2f3b91003011109n2d817de9v96b491b40eacae32@mail.gmail.com>
Message-ID: <7c74f5761003020328u3a480d59v703899d3fc81a1fb@mail.gmail.com>

Thanks Michael right now it's working correct.
In wiki example there is a lack of args "tts_engine" "tts_voice" and
"max_timeouts"

BR,
Micha?

2010/3/1 Michael Collins <msc at freeswitch.org>:
>
>
> 2010/3/1 michal kalinowski <michal.kalinowski at interia.pl>
>>
>> Thank's Brian
>>
>> So at this moment i have only clean configuration in lua without any XML.
>>
>> I get example from wiki:
>>
>> hash={["main"]=undef,
>>
>> ? ["name"]="top",
>>
>> ? ? ? ? ["greet_long"]="phrase:demo_ivr_main_menu",
>>
>> ["greet_short"]="phrase:demo_ivr_main_menu_short",
>>
>> ["invalid_sound"]="ivr/ivrthat_was_an_invalid_entry.wav",
>>
>> ["exit_sound"]="voicemail/vmgoodbye.wav",
>>
>> ["confirm_macro"]="undef",
>>
>> ?["confirm_key"]="undef",
>>
>> ? ?["confirm_attempts"]="3",
>>
>> ? ? ?["inter_digit_timeout"]="2000",
>>
>> ? ? ? ["digit_len"]="1",
>>
>> ? ? ? ? ? ? ? ["timeout"]="10000",
>> ["max_failures"]="3"
>> }
>> top = freeswitch.IVRMenu(hash["main"],
>> hash["name"],
>> hash["greet_long"],hash["greet_short"],
>> hash["invalid_sound"],
>> hash["exit_sound"],
>> hash["confirm_macro"],
>> hash["confirm_key"],
>> hash["confirm_attempts"],
>> hash["inter_digit_timeout"],
>> hash["digit_len"],
>> hash["timeout"],
>> hash["max_failures"]);
>>
>> top:bindAction("menu-exec-app", "playback/tmp/swimp.raw","2");
>> top:execute(session, "top");
>>
>> But something is wrong freeswitch say
>> 2010-03-01 13:02:43.633721 [ERR] mod_lua.cpp:182 Error in IVRMenu
>> expected 16..16 args, got 13
>> stack traceback:
>> ? ? ? ?[C]: in function 'IVRMenu'
>> ? ? ? ?/usr/local/freeswitch/scripts/ivr.lua:19: in main chunk
>>
>> What args FS expect?
>
> I ran into this issue when I was converting the demo IVR to Lua. There are
> indeed 16 arguments but you've only got 13 of them. Here's a snippet from
> the Lua script I'm working on:
> ivr_def = {
> ??? ["main"]??????????????? = undef,
> ??? ["name"]??????????????? = "demo_ivr_lua",
> ??? ["greet_long"]????????? = "phrase:demo_ivr_main_menu",
> ??? ["greet_short"]???????? = "phrase:demo_ivr_main_menu_short",
> ??? ["invalid_sound"]?????? = "ivr/ivr-that_was_an_invalid_entry.wav",
> ??? ["exit_sound"]????????? = "voicemail/vm-goodbye.wav",
> ??? ["confirm_macro"]?????? = "",
> ??? ["confirm_key"]???????? = "",
> ??? ["tts_engine"]????????? = "flite",
> ??? ["tts_voice"]?????????? = "rms",
> ??? ["confirm_attempts"]??? = "3",
> ??? ["inter_digit_timeout"] = "2000",
> ??? ["digit_len"]?????????? = "4",
> ??? ["timeout"]???????????? = "10000",
> ??? ["max_failures"]??????? = "3",
> ??? ["max_timeouts"]??????? = "2"
> }
> Those are all 16 arguments. Notice you need tts_engine, tts_voice, and
> max_timeouts. Try that and let us know how it goes.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From yehavi.bourvine at gmail.com  Tue Mar  2 04:56:53 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 2 Mar 2010 14:56:53 +0200
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
	<191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
Message-ID: <d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>

Since I am aware that I am the only one who experience this problem, I am
trying to ivestigate it, although I do not know the source code well...

I removed all the datbase calls from our scripts and the problem still
persists. It happens usually when two calls arrive to the same extension
(usually to an SLA one) and one of them disconnects during the ringing
phase.

I've inserted some "printfs" in switch_core_sqldb.c and found that when this
happens the variable dbh->io_mutex has a null value, thus no mutex is locked
(as far as I understand it).

I hope this might give some clues.

                      Thanks! __Yehavi:

 2010/3/1 Anthony Minessale <anthony.minessale at gmail.com>

>  You probably had it right in the last email.
> Polling the core's db from lua is not recommended.
> We use sla extensively and never once see your issue.
>
>  On Feb 28, 2010 8:55 AM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
>  After a few more tests I *think* it is related to SLA. Since it cannot be
> reproduced consistenty I say "think".
>
> What I have is:
> - 80635 which is an SLA extension between two Polycoms.
> - 80636 which is a private extension on Polycom
> - 80632 which is a Cisco extension.
> - 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this
> gateway doesn't hav any login information it is not defined as a gateway but
> accepted via ACL.
>
> The tests that passed ok:
> - From/to 80636/80632 to 80635 (i.e. - all internal).
> - From/to 80635 to 80636/80632 (i.e. - all internal)
> - From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single
> extensions).
>
> The one that fails after a few attempts:
> - From 86111 to 80635 (i.e. from the gateway to a shared extension).
>
> I hope that this gives some more clues.
>
>                           Thanks, __Yehavi:
>
>  2010/2/28 Yehavi Bourvine <yehavi.bourvine at gmail.com>
>
>
> >
> > Hello,
> >
> >   I've installed a machine with CentOS-5.4 and the latest FreeSwitch
> (16841M). The...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dftoro at yahoo.com  Tue Mar  2 05:24:59 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 2 Mar 2010 05:24:59 -0800 (PST)
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
Message-ID: <251258.58331.qm@web33508.mail.mud.yahoo.com>

Hi Phillip Jones,

Any concerns, You tell me, I help you

Diego Toro
http://lacarretade.blogspot.com/


--- On Mon, 3/1/10, Phillip Jones <pjintheusa at gmail.com> wrote:

> From: Phillip Jones <pjintheusa at gmail.com>
> Subject: Re: [Freeswitch-users] .NET / ESL
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, March 1, 2010, 10:01 PM
> Thanks
> 
> On Mon, Mar 1, 2010 at 9:42 PM,
> Brian West <brian at freeswitch.org>
> wrote:
> 
> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
> 
> 
> 
> in the managed folder.
> 
> 
> 
> /b
> 
> 
> 
> On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
> 
> 
> 
> > Hi there,
> 
> >
> 
> > I am sure that I have read somewhere that someone is
> developing a new .NET library for FreeSWITCH ESL. If so can
> someone point me to some info / code etc. so I can take a
> look.
> 
> >
> 
> > Thanks
> 
> >
> 
> > pj
> 
> >
> _______________________________________________
> 
> > FreeSWITCH-users mailing list
> 
> > FreeSWITCH-users at lists.freeswitch.org
> 
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> > http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> 
> FreeSWITCH-users mailing list
> 
> FreeSWITCH-users at lists.freeswitch.org
> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> 
> http://www.freeswitch.org
> 
> 
> 
> 
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      



From infos at madovsky.org  Tue Mar  2 06:25:54 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 09:25:54 -0500
Subject: [Freeswitch-users] Freeswitch & T.38
References: <1267528831989-4660389.post@n2.nabble.com>
Message-ID: <4BB20F0B3FCB43DCAD902617DDA79EEC@MOBILEE1705>

google wiki media proxy freeswitch

----- Original Message ----- 
From: "Sharad" <sharad at coraltele.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 02, 2010 6:20 AM
Subject: [Freeswitch-users] Freeswitch & T.38


>
> Hi friends,
>
> Is Freeswitch 1.05 carrying T.38 support also ?
>
> If not, do we have any community member who can provide paid T.38 support
> with Freeswitch ?
>
>
> -- 
> View this message in context: 
> http://n2.nabble.com/Freeswitch-T-38-tp4660389p4660389.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 




From brian at freeswitch.org  Tue Mar  2 06:28:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:28:45 -0600
Subject: [Freeswitch-users] wiki broken link pocketsphinx
In-Reply-To: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
Message-ID: <5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>

No thats not right either.  Care to elaborate?

/b

On Mar 2, 2010, at 12:05 AM, Madovsky wrote:

> Hello,
>  
> the link that goes to grammar sphinx is broken.
> the right link is
>  
> http://cmusphinx.sourceforge.net/sphinx4/
>  
> Regards
>  
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar  2 06:32:05 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:32:05 -0600
Subject: [Freeswitch-users] SQL ERR: library routine called out of
	sequence
In-Reply-To: <d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>
References: <d361ca7b1002160656w7d616f6dn5e80087ebde94dbe@mail.gmail.com>
	<191c3a031002160755q569fe2ednfefb01b6d9cf7f0e@mail.gmail.com>
	<d361ca7b1002161059w1ef9484dr87e69a33aa2b62fc@mail.gmail.com>
	<191c3a031002161110s637a48a2r717ae94557fe9e9a@mail.gmail.com>
	<d361ca7b1002161130k3eb2d88by3100b30a9d29bd51@mail.gmail.com>
	<191c3a031002161230h2b6e6f9wf3f65ddb71f08297@mail.gmail.com>
	<d361ca7b1002280514p71003588t8a0daebce5ba9e4d@mail.gmail.com>
	<d361ca7b1002280647l700206cfv6e130351d8ac6d5@mail.gmail.com>
	<191c3a031002282105p4ed80135jc115a007ec7e0d4a@mail.gmail.com>
	<191c3a031002282106t59801ddctb12c4fab160e9a07@mail.gmail.com>
	<d361ca7b1003020456j3876ce32j8418cdcbd52b6ea5@mail.gmail.com>
Message-ID: <0ADA063F-9E3E-4540-BB3B-48373211BD6A@freeswitch.org>

Yes accessing the db in the method you're doing is unsupport and not recommended.

/b

On Mar 2, 2010, at 6:56 AM, Yehavi Bourvine wrote:

> Since I am aware that I am the only one who experience this problem, I am trying to ivestigate it, although I do not know the source code well...
>  
> I removed all the datbase calls from our scripts and the problem still persists. It happens usually when two calls arrive to the same extension (usually to an SLA one) and one of them disconnects during the ringing phase.
>  
> I've inserted some "printfs" in switch_core_sqldb.c and found that when this happens the variable dbh->io_mutex has a null value, thus no mutex is locked (as far as I understand it).
>  
> I hope this might give some clues.
>  
>                       Thanks! __Yehavi:




From brian at freeswitch.org  Tue Mar  2 06:33:59 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:33:59 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable destination
In-Reply-To: <23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
Message-ID: <074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>

Yes that is because its @30i, All codecs offered MUST be 30ms if you want to offer iLBC at 30i.  So add @30i to all but GSM since you can't do GSM @ 30.

/b
On Mar 2, 2010, at 1:01 AM, Seven Du wrote:

> sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)
> 
> 
> 2010/3/2 Seven Du <dujinfang at gmail.com>:
>> I have iLBC configured in profile internal
>> 
>> CODECS IN               PCMU,PCMA,iLBC at 30i,GSM
>> CODECS OUT              PCMU,PCMA,iLBC at 30i,GSM
>> 
>> and     <param name="inbound-codec-negotiation" value="generous"/>
>> 
>> but it seems not comparing with iLBC
>> 
>>  010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
>> payload to 101
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>> 
>> 
>> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>> 
>> Thanks.
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Mar  2 06:34:19 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 08:34:19 -0600
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
Message-ID: <FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>

I have called netgear worse.  :)

/b

On Mar 2, 2010, at 3:03 AM, Tihomir Culjaga wrote:

> bastard !!!! :)
> 
> T.




From jonas.gauffin at gmail.com  Tue Mar  2 07:02:47 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 16:02:47 +0100
Subject: [Freeswitch-users] Incorrect nonce
Message-ID: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>

Hello,

I got a problem with rejected calls due to 403 (happens sometimes).
My sip provider found the problem, the nonce used by FS is sometimes
incorrect (I got the trace from my sip provider).

A sip trace can be found here: http://pastebin.freeswitch.org/12280


2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
(sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_INIT
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
(sofia/external/0707728XXX) State INIT
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX
SOFIA INIT
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
(sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
(sofia/external/0707728XXX) State INIT going to sleep
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
(sofia/external/0707728XXX) State ROUTING
2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX
SOFIA ROUTING
2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
(sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
sofia/external/0707728XXX [BREAK]
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
(sofia/external/0707728XXX) State ROUTING going to sleep
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
(sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
(sofia/external/0707728XXX) State CONSUME_MEDIA
2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
(sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][0]
2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][0]
2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][100]
2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [calling][100]
2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
sofia/external/0707728XXX entering state [terminated][407]
2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]

Regards,
  Jonas
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From m.sobkow at marketelsystems.com  Tue Mar  2 07:06:15 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 09:06:15 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <20100301221524.GB1751@hijacked.us>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
Message-ID: <4B8D2967.7010403@marketelsystems.com>

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From infos at madovsky.org  Tue Mar  2 07:09:38 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 10:09:38 -0500
Subject: [Freeswitch-users] wiki broken link pocketsphinx
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
	<5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>
Message-ID: <30970655E0CD476A95CF8382A2742084@MOBILEE1705>

apparently Diego updated it.
I don't know if I'm skilled enough to elaborate

  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 9:28 AM
  Subject: Re: [Freeswitch-users] wiki broken link pocketsphinx


  No thats not right either.  Care to elaborate?


  /b


  On Mar 2, 2010, at 12:05 AM, Madovsky wrote:


    Hello,

    the link that goes to grammar sphinx is broken.
    the right link is

    http://cmusphinx.sourceforge.net/sphinx4/

    Regards

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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tomb at cachecomm.com  Tue Mar  2 07:15:50 2010
From: tomb at cachecomm.com (Tom)
Date: Tue, 02 Mar 2010 08:15:50 -0700
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
References: <4B8BEE7F.1030401@cachecomm.com>	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>	<4B8C273C.6010808@cachecomm.com>	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>	<4B8C5AAE.2030908@cachecomm.com>	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
	<FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
Message-ID: <4B8D2BA6.1030907@cachecomm.com>

Brian West wrote:
> I have called netgear worse.  :)
>
> /b
>
> On Mar 2, 2010, at 3:03 AM, Tihomir Culjaga wrote:
>
>   
>> bastard !!!! :)
>>
>> T.
>>     
>
>
>   
I removed the Netgear P.O.S . and put the phones behind a linux firewall 
all is good.
Thanks for the help

Tom

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




From m.sobkow at marketelsystems.com  Tue Mar  2 07:39:20 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 09:39:20 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8D2967.7010403@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
	<4B8D2967.7010403@marketelsystems.com>
Message-ID: <4B8D3128.6020807@marketelsystems.com>

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From moises.silva at gmail.com  Tue Mar  2 07:45:22 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Mar 2010 10:45:22 -0500
Subject: [Freeswitch-users] Freeswitch/Openzap dials out on disconnected
	FXO line
In-Reply-To: <4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
References: <Acq2eHpz1O0wasTJQ1GGlQN3irPlFA==>
	<4DF42CB92831454193CEC0E375E06725@greyhawk.tonecommander.com>
Message-ID: <c4d05cbe1003020745h20a1fdccmc7486c6454b810d5@mail.gmail.com>

Please try latest openzap (svn update in libs/openzap). I commited some code
that may help you.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Thu, Feb 25, 2010 at 7:13 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  When dialing out, Freeswitch/Openzap is not detecting that an analog FXO
> channel is disconnected and tries dialing out on the channel anyway.  No
> error is reported.  The call doesn?t timeout until a minute later.
> Shouldn?t Freeswitch/Openzap skip over a disconnected channel to the next
> connected channel?
>
>
>
> I have configured a Sangoma A200 FXO card as a FXO span.
>
>
>
> [span wanpipe FXO]
>
> name => PSTN Line 1
>
> number => 4253491059
>
> fxo-channel => 2:3
>
> name => PSTN Line 2
>
> number => 4253491058
>
> fxo-channel => 2:4
>
>
>
>
>
> The wanpipe driver does detect and report when a CO line is connected or
> disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as
> reported in the log.
>
>
>
> /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3:
> FXO Line is disconnected!
>
> FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_TRAP][3:1] STATE [DOWN]
>
>
>
> /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4:
> FXO Line is connected!
>
> FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT
> [ALARM_CLEAR][3:2] STATE [DOWN]
>
>
>
>
>
> I have the dialplan configured to use the next available port in the FXO
> span (there will be more than 2 channels later).
>
>
>
> <extension name="OutgoingFXO">
>
>   <condition field="destination_number" expression="^7(\d+)$">
>
>     <action application="bridge" data="openzap/FXO/a/$1"/>
>
>   </condition>
>
> </extension>
>
>
>
>
>
> Here is a portion of the log when that shows dialing out on a disconnected
> analog FXO channel.
>
>
>
> EXECUTE sofia/internal/5410 at 192.168.72.45:5060bridge(openzap/FXO/a/93491045)
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound
> channel OpenZAP/3:1/93491045
>
> 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel
> OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7]
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269
> (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1
> from DOWN to DIALING
>
> 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486
> OpenZAP/3:1/93491045 does not support the proxy feature, disabling.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread
> starting.
>
> 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state
> handler on 3:1 for DIALING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045)
> State Change CS_INIT -> CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338
> (OpenZAP/3:1/93491045) State INIT going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045
> CHANNEL ROUTING
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66
> (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal
> OpenZAP/3:1/93491045 [BREAK]
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341
> (OpenZAP/3:1/93491045) State ROUTING going to sleep
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314
> (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA
>
> 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360
> (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From scottferri09 at gmail.com  Tue Mar  2 07:53:09 2010
From: scottferri09 at gmail.com (Scott Fernandez)
Date: Tue, 2 Mar 2010 21:23:09 +0530
Subject: [Freeswitch-users] Inbound settings
Message-ID: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>

Hi,


     1) How to configure inbound settings in FS?
     2) If so, Please list down the steps that I require to do?
     3) If have more than 1 DID number (let;'s say 4 DID's
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From brian at freeswitch.org  Tue Mar  2 08:00:22 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 10:00:22 -0600
Subject: [Freeswitch-users] Inbound settings
In-Reply-To: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
References: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
Message-ID: <D9251A3E-3341-4928-84DB-FE242DE6CAAC@freeswitch.org>

Have you read over the wiki?  http://wiki.freeswitch.org/wiki/Getting_Started_Guide

/b

On Mar 2, 2010, at 9:53 AM, Scott Fernandez wrote:

> Hi,
>  
>  
>      1) How to configure inbound settings in FS?
>      2) If so, Please list down the steps that I require to do?
>      3) If have more than 1 DID number (let;'s say 4 DID's
>  
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From infos at madovsky.org  Tue Mar  2 08:02:12 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:02:12 -0500
Subject: [Freeswitch-users] Inbound settings
References: <df97b9c51003020753p75d2b7c8l45c2a9d42efd3357@mail.gmail.com>
Message-ID: <860AD4D8E1F34919A28DF973FF2C7969@MOBILEE1705>

http://wiki.freeswitch.org/wiki/Installation_Guide

  ----- Original Message ----- 
  From: Scott Fernandez 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 10:53 AM
  Subject: [Freeswitch-users] Inbound settings


  Hi,


       1) How to configure inbound settings in FS?
       2) If so, Please list down the steps that I require to do?
       3) If have more than 1 DID number (let;'s say 4 DID's




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From m.sobkow at marketelsystems.com  Tue Mar  2 08:10:43 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 02 Mar 2010 10:10:43 -0600
Subject: [Freeswitch-users] How to tell if a call timed out
In-Reply-To: <4B8D3128.6020807@marketelsystems.com>
References: <4B8C32D7.40202@marketelsystems.com>
	<20100301221524.GB1751@hijacked.us>
	<4B8D2967.7010403@marketelsystems.com>
	<4B8D3128.6020807@marketelsystems.com>
Message-ID: <4B8D3883.9040902@marketelsystems.com>

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From sos at sokhapkin.dyndns.org  Tue Mar  2 08:13:06 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 11:13:06 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
Message-ID: <201003021113.06820.sos@sokhapkin.dyndns.org>

FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No single 
SIP request hits it. What does it doing? I unloaded all modules which can be 
unloaded, no difference. Tried with both -vm and -heavy-timer start up 
options. I'm running gentoo linux.

# uname -a
Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686 Intel(R) 
Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux



From jonas.gauffin at gmail.com  Tue Mar  2 08:20:26 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 17:20:26 +0100
Subject: [Freeswitch-users] Diversion header
Message-ID: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>

Hello,

I got a incoming call from my provider which goes to a javascript. The
javascript transfers the call to an external destination (goes back through
my provider).
A diversion headers is added by FreeSWITCH on the outbound call. How can I
remove it (or if not possible, how do I change it)?

javascript:
        session.setCallerData("caller_id_number", aNumber);
session.setVariable("gate_caller_id_number", aNumber);
        session.setVariable("gate_site_id", SITE_ID);
        session.execute("transfer", this.dtmfBuffer + " XML internal");



send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200:
   ------------------------------------------------------------------------
   INVITE sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>SIP/2.0
   Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS
   Max-Forwards: 67
   From: "0236661XXX" <sip:0243795362 at 212.247.XX.XX>;tag=BHvytKtv47r2a
   To: <sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>
>
   Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5
   CSeq: 127654862 INVITE
   Contact: <sip:gw+dalaconnecttele2 at 212.247.XX.XX
:5060;transport=udp;gw=dalaconnecttele2>
   User-Agent: Gateon
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 273
   Diversion: 0 <sip:0 at 212.151.Y.Y
;user=phone>;reason=unconditional;counter=1;privacy=off
   X-FS-Support: update_display
   Remote-Party-ID: "0236661XXX" <sip:0243795362 at 212.247.XX.XX
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX
   s=FreeSWITCH
   c=IN IP4 212.247.XX.XX
   t=0 0
   m=audio 32712 RTP/AVP 8 3 101 13
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
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From infos at madovsky.org  Tue Mar  2 08:20:54 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:20:54 -0500
Subject: [Freeswitch-users] last svn version compile error
Message-ID: <C8664F4269084354BB7AB9203887E30E@MOBILEE1705>

The compilation is fine if I comment out mod_vmd from modules.conf

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 1:24 AM
  Subject: last svn version compile error


  Hi,

  Ino succes to update freeswitch to version 16859
  with usual command "make current" etc...
  but now I got this error :

  /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of ?switch_core_media_bug_add? makes integer from pointer without a cast
  /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function ?switch_core_media_bug_add?

  Any idea ?

  thanks

  Franck

  Linux Fedora 10  64bits
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From moises.silva at gmail.com  Tue Mar  2 08:35:47 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Mar 2010 11:35:47 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>

ps -LlFm -p `pidof freeswitch`

That shows you CPU usage per FreeSWITCH thread

Then

pstack `pidof freeswitch`

That shows the stack trace of every FreeSWITCH thread and then you can find
exactly what is doing.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Tue, Mar 2, 2010 at 11:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> single
> SIP request hits it. What does it doing? I unloaded all modules which can
> be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Mar  2 08:42:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 10:42:56 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>

If you don't like that CPU usage, use a different kernel or disable the
broken "tickles" features.

you can't use both -vm and -heavy-timer because -heavy-timer replaces -vm
permanently.


On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> single
> SIP request hits it. What does it doing? I unloaded all modules which can
> be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  2 08:43:45 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:43:45 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <6FB1E9CF2C044BBCBE87D44335CD2383@MOBILEE1705>

Hi,

I searched some articles of FS in a cluster environment
but there are some unclear writes :

in http://www.technoclicks.com/article-2545.php from 2006 it says

"Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

and this thread from 2008
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
that describes contradiction of how to manage clustering.

so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

Regards

F
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From anthony.minessale at gmail.com  Tue Mar  2 08:44:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 10:44:23 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
Message-ID: <191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>

Are you reporting what you think is a bug on the mailing list?
I think you have been around long enough to know better.............


On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes
> incorrect (I got the trace from my sip provider).
>
> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>
>
> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_INIT
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/0707728XXX) State INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX
> SOFIA INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/0707728XXX) State INIT going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/0707728XXX) State ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
> sofia/external/0707728XXX SOFIA ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
> sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/0707728XXX) State ROUTING going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/0707728XXX) State CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
> sofia/external/0707728XXX entering state [terminated][407]
> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>
> Regards,
>   Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar  2 08:53:35 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 10:53:35 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
Message-ID: <151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>

Jonas,
	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.

Remember the list is not a bug tracker... Thats what we have jira for  ;)

Thanks,
Brian

On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:

> Are you reporting what you think is a bug on the mailing list?
> I think you have been around long enough to know better.............
> 
> 
> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> Hello,
> 
> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
> 
> A sip trace can be found here: http://pastebin.freeswitch.org/12280
> 
> 
> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
> 
> Regards,
>   Jonas
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Tue Mar  2 08:56:52 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 11:56:52 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <76B2210A926D4D06B34B38BBE633210D@MOBILEE1705>

what FS team thinks about this SIP clustering architecture ?

http://blogs.sun.com/kshitiz/entry/converged_load_balancer

Regards

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 11:43 AM
  Subject: clustering question


  Hi,

  I searched some articles of FS in a cluster environment
  but there are some unclear writes :

  in http://www.technoclicks.com/article-2545.php from 2006 it says

  "Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

  and this thread from 2008
  http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
  that describes contradiction of how to manage clustering.

  so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

  Regards

  F
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From sos at sokhapkin.dyndns.org  Tue Mar  2 09:00:15 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 12:00:15 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<191c3a031003020842t53fc0906qea1c7c19d4fd904b@mail.gmail.com>
Message-ID: <201003021200.16252.sos@sokhapkin.dyndns.org>

Tickless feature is disabled in the kernel.

$ zcat /proc/config.gz | grep CONFIG_NO_HZ
# CONFIG_NO_HZ is not set


On Tuesday 02 March 2010, Anthony Minessale wrote:
> If you don't like that CPU usage, use a different kernel or disable the
> broken "tickles" features.
>
> you can't use both -vm and -heavy-timer because -heavy-timer replaces -vm
> permanently.
>
>
> On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No
> > single
> > SIP request hits it. What does it doing? I unloaded all modules which can
> > be
> > unloaded, no difference. Tried with both -vm and -heavy-timer start up
> > options. I'm running gentoo linux.
> >
> > # uname -a
> > Linux  2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686
> > Intel(R)
> > Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From jonas.gauffin at gmail.com  Tue Mar  2 09:04:26 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 18:04:26 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
Message-ID: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>

You are both correct.

Issue created.

On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:

> Jonas,
> Also I need a complete pcap of this without modifications.  I can't hand
> calculate this to see what is going on without intact data. Please open a
> jira and email me the pcap in private if you must.
>
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>
> Thanks,
> Brian
>
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>
> Are you reporting what you think is a bug on the mailing list?
> I think you have been around long enough to know better.............
>
>
> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello,
>>
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes
>> incorrect (I got the trace from my sip provider).
>>
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>
>>
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>> (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>> sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>> (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>> (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>> sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>> sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>> (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>> sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>
>> Regards,
>>   Jonas
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar  2 09:06:20 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 12:06:20 -0500
Subject: [Freeswitch-users] clustering question
Message-ID: <C3B965932B59454EAC61E6EC90766510@MOBILEE1705>

also from 2007
http://ftp.iptel.org/pub/sems/fostel_2007_coeffic_sems_load_balancing.pdf
that says SEMS can make round robin and hash load balancing.
I don't talk here about failover since this can be resolved with conventional clustering solution
as hearbeat pacemaker redhat cluster etc... but especially load balancing to increase nujmber
of calls

Regards

F


  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 11:56 AM
  Subject: Re: clustering question


  what FS team thinks about this SIP clustering architecture ?

  http://blogs.sun.com/kshitiz/entry/converged_load_balancer

  Regards

  F
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 02, 2010 11:43 AM
    Subject: clustering question


    Hi,

    I searched some articles of FS in a cluster environment
    but there are some unclear writes :

    in http://www.technoclicks.com/article-2545.php from 2006 it says

    "Minessale says FreeSwitch won't be bounded by those same problems. The software is designed to run on a cluster of an unlimited number of servers though realistically that will be gated by the capacity of the network. Early tests on a single server showed simultaneous calls on a Pentium server running at . GHz. The server completed calls per second with bursts to calls per second."

    and this thread from 2008
    http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg05258.html
    that describes contradiction of how to manage clustering.

    so is FS really be built as cluster to expand call capacity ? or is it be clustered by the fact that only serveral FS share a DB ?

    Regards

    F
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From jonas.gauffin at gmail.com  Tue Mar  2 09:35:33 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 18:35:33 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
Message-ID: <d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>

I solved it by adding '<action application="set" data="sip_h_Diversion="/>'
to my generated dial plan. But why was the header added in the first place?

On Tue, Mar 2, 2010 at 5:20 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> I got a incoming call from my provider which goes to a javascript. The
> javascript transfers the call to an external destination (goes back through
> my provider).
> A diversion headers is added by FreeSWITCH on the outbound call. How can I
> remove it (or if not possible, how do I change it)?
>
> javascript:
>         session.setCallerData("caller_id_number", aNumber);
> session.setVariable("gate_caller_id_number", aNumber);
>         session.setVariable("gate_site_id", SITE_ID);
>         session.execute("transfer", this.dtmfBuffer + " XML internal");
>
>
>
> send 1179 bytes to udp/[130.244.Y.YY]:5060 at 16:13:16.752200:
>    ------------------------------------------------------------------------
>    INVITE sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>SIP/2.0
>    Via: SIP/2.0/UDP 212.247.XX.XX;rport;branch=z9hG4bKaa59XjFt2N2rS
>    Max-Forwards: 67
>    From: "0236661XXX" <sip:0243795362 at 212.247.XX.XX>;tag=BHvytKtv47r2a
>    To: <sip:024390510 at sip-corporate.provider.com<sip%3A024390510 at sip-corporate.provider.com>
> >
>    Call-ID: 5a022f8f-a0b9-122d-779a-dbdeced2caa5
>    CSeq: 127654862 INVITE
>    Contact: <sip:gw+dalaconnecttele2 at 212.247.XX.XX
> :5060;transport=udp;gw=dalaconnecttele2>
>    User-Agent: Gateon
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 273
>    Diversion: 0 <sip:0 at 212.151.Y.Y
> ;user=phone>;reason=unconditional;counter=1;privacy=off
>    X-FS-Support: update_display
>    Remote-Party-ID: "0236661XXX" <sip:0243795362 at 212.247.XX.XX
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1267513684 1267513685 IN IP4 212.247.XX.XX
>    s=FreeSWITCH
>    c=IN IP4 212.247.XX.XX
>    t=0 0
>    m=audio 32712 RTP/AVP 8 3 101 13
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>
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From msc at freeswitch.org  Tue Mar  2 09:40:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 09:40:56 -0800
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
Message-ID: <87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>

On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy
<lakindia89 at gmail.com>wrote:

> Hi all,
> I did make current from my freeswitch src directory, but the system got
> hanged.
> Any one know why this happened.
>

Does it happen every time you make current? If so, does it always happen at
the "expat" lib?
-MC
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From msc at freeswitch.org  Tue Mar  2 09:44:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 09:44:16 -0800
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <251258.58331.qm@web33508.mail.mud.yahoo.com>
References: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
	<251258.58331.qm@web33508.mail.mud.yahoo.com>
Message-ID: <87f2f3b91003020944s77e57eavc5e8e503c0c15cbb@mail.gmail.com>

On Tue, Mar 2, 2010 at 5:24 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi Phillip Jones,
>
> Any concerns, You tell me, I help you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>

Diego, I have a concern: you're supposed to join the FS community conf call
and talk to us about using .NET langs with ESL! :P

Seriously, though, we would love to have you join the conference tomorrow or
any other Wednesday and discuss what's going on with .NET ESL programming.
Moc has a server you can use to share a screen if you'd like.

Thanks!!

-MC
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From brian at freeswitch.org  Tue Mar  2 09:45:27 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 11:45:27 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
Message-ID: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>

Because it should be there if you redirected the call.  

/b

On Mar 2, 2010, at 11:35 AM, Jonas Gauffin wrote:

> I solved it by adding '<action application="set" data="sip_h_Diversion="/>' to my generated dial plan. But why was the header added in the first place?




From brian at freeswitch.org  Tue Mar  2 09:47:39 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 11:47:39 -0600
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
	<87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
Message-ID: <2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>

what os are you on?

/b

On Mar 2, 2010, at 11:40 AM, Michael Collins wrote:

> 
> 
> On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy <lakindia89 at gmail.com> wrote:
> Hi all,
> I did make current from my freeswitch src directory, but the system got hanged.
> Any one know why this happened.
> 
> Does it happen every time you make current? If so, does it always happen at the "expat" lib?
> -MC
> 
> _________

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From sos at sokhapkin.dyndns.org  Tue Mar  2 09:51:48 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 12:51:48 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
Message-ID: <201003021251.48760.sos@sokhapkin.dyndns.org>

FS spends half of CPU here (main thread):

#0  0xffffe424 in __kernel_vsyscall ()
#1  0xb7a82781 in select () from /lib/libc.so.6
#2  0xb7e392eb in apr_sleep () from /opt/freeswitch/lib/libfreeswitch.so.1
#3  0x00000000 in ?? ()

The second half is spent here:

#0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
#1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
#2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8, 
obj=0xb7101da8) at src/switch_loadable_module.c:96
#3  0xb7e379a8 in dummy_worker () from /opt/freeswitch/lib/libfreeswitch.so.1


On Tuesday 02 March 2010, Moises Silva wrote:
> ps -LlFm -p `pidof freeswitch`
>
> That shows you CPU usage per FreeSWITCH thread
>
> Then
>
> pstack `pidof freeswitch`
>
> That shows the stack trace of every FreeSWITCH thread and then you can find
> exactly what is doing.





From anthony.minessale at gmail.com  Tue Mar  2 10:00:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:00:23 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021251.48760.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c4d05cbe1003020835t7dfa0508q6b67579da21c3a1f@mail.gmail.com>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>

Well, there you go.
your box is having a hard time doing a simple 1ms sleep in a loop for the
centralized core timer.

It's either your kernel or your box.  My box and every box I have do not
have that problem so you may want to consider a new one.

FreeSWITCH is FREE so give it a good home.

The good news is, you can make upwards of 1500 calls on that same 2% of cpu
usage.



On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> FS spends half of CPU here (main thread):
>
> #0  0xffffe424 in __kernel_vsyscall ()
> #1  0xb7a82781 in select () from /lib/libc.so.6
> #2  0xb7e392eb in apr_sleep () from /opt/freeswitch/lib/libfreeswitch.so.1
> #3  0x00000000 in ?? ()
>
> The second half is spent here:
>
> #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> obj=0xb7101da8) at src/switch_loadable_module.c:96
> #3  0xb7e379a8 in dummy_worker () from
> /opt/freeswitch/lib/libfreeswitch.so.1
>
>
> On Tuesday 02 March 2010, Moises Silva wrote:
> > ps -LlFm -p `pidof freeswitch`
> >
> > That shows you CPU usage per FreeSWITCH thread
> >
> > Then
> >
> > pstack `pidof freeswitch`
> >
> > That shows the stack trace of every FreeSWITCH thread and then you can
> find
> > exactly what is doing.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jonas.gauffin at gmail.com  Tue Mar  2 10:11:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 19:11:54 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
Message-ID: <d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>

Sure I can buy that, but not with a URI that doesnt match the inbound call.

"0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore all
calls through that javascript will be rejected by my provider.


On Tue, Mar 2, 2010 at 6:45 PM, Brian West <brian at freeswitch.org> wrote:

> Because it should be there if you redirected the call.
>
> /b
>
> On Mar 2, 2010, at 11:35 AM, Jonas Gauffin wrote:
>
> > I solved it by adding '<action application="set"
> data="sip_h_Diversion="/>' to my generated dial plan. But why was the header
> added in the first place?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sos at sokhapkin.dyndns.org  Tue Mar  2 10:16:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 13:16:30 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
Message-ID: <201003021316.30842.sos@sokhapkin.dyndns.org>

I see the same problem on 4 absolutely different boxes with different linux 
kernels.

On Tuesday 02 March 2010, Anthony Minessale wrote:
> Well, there you go.
> your box is having a hard time doing a simple 1ms sleep in a loop for the
> centralized core timer.
>
> It's either your kernel or your box.  My box and every box I have do not
> have that problem so you may want to consider a new one.
>
> FreeSWITCH is FREE so give it a good home.
>
> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
> usage.
>
>
>
> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>
> <sos at sokhapkin.dyndns.org>wrote:
> > FS spends half of CPU here (main thread):
> >
> > #0  0xffffe424 in __kernel_vsyscall ()
> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > #2  0xb7e392eb in apr_sleep () from
> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> >
> > The second half is spent here:
> >
> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > #3  0xb7e379a8 in dummy_worker () from
> > /opt/freeswitch/lib/libfreeswitch.so.1
> >
> > On Tuesday 02 March 2010, Moises Silva wrote:
> > > ps -LlFm -p `pidof freeswitch`
> > >
> > > That shows you CPU usage per FreeSWITCH thread
> > >
> > > Then
> > >
> > > pstack `pidof freeswitch`
> > >
> > > That shows the stack trace of every FreeSWITCH thread and then you can
> >
> > find
> >
> > > exactly what is doing.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From brian at freeswitch.org  Tue Mar  2 10:16:48 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 12:16:48 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
Message-ID: <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>

Was the header there on the inbound invite?

/b

On Mar 2, 2010, at 12:11 PM, Jonas Gauffin wrote:

> Sure I can buy that, but not with a URI that doesnt match the inbound call.
> 
> "0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore all calls through that javascript will be rejected by my provider.
> 
> 

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From gmaruzz at celliax.org  Tue Mar  2 10:23:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 2 Mar 2010 19:23:04 +0100
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021316.30842.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org> 
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com> 
	<201003021316.30842.sos@sokhapkin.dyndns.org>
Message-ID: <7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>

Sergey,

that's not a problem, is how FS works.

That load on the CPU when idle is actually not important. Is just the
minimum, like in a carburettor.

If you put it under load, you will see that there will need some
serious load (many calls) to have it ramp up the cpu usage.

-giovanni

On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> I see the same problem on 4 absolutely different boxes with different linux
> kernels.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
>> Well, there you go.
>> your box is having a hard time doing a simple 1ms sleep in a loop for the
>> centralized core timer.
>>
>> It's either your kernel or your box. ?My box and every box I have do not
>> have that problem so you may want to consider a new one.
>>
>> FreeSWITCH is FREE so give it a good home.
>>
>> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
>> usage.
>>
>>
>>
>> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org>wrote:
>> > FS spends half of CPU here (main thread):
>> >
>> > #0 ?0xffffe424 in __kernel_vsyscall ()
>> > #1 ?0xb7a82781 in select () from /lib/libc.so.6
>> > #2 ?0xb7e392eb in apr_sleep () from
>> > /opt/freeswitch/lib/libfreeswitch.so.1 #3 ?0x00000000 in ?? ()
>> >
>> > The second half is spent here:
>> >
>> > #0 ?0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
>> > #1 ?0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
>> > #2 ?0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
>> > obj=0xb7101da8) at src/switch_loadable_module.c:96
>> > #3 ?0xb7e379a8 in dummy_worker () from
>> > /opt/freeswitch/lib/libfreeswitch.so.1
>> >
>> > On Tuesday 02 March 2010, Moises Silva wrote:
>> > > ps -LlFm -p `pidof freeswitch`
>> > >
>> > > That shows you CPU usage per FreeSWITCH thread
>> > >
>> > > Then
>> > >
>> > > pstack `pidof freeswitch`
>> > >
>> > > That shows the stack trace of every FreeSWITCH thread and then you can
>> >
>> > find
>> >
>> > > exactly what is doing.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From anthony.minessale at gmail.com  Tue Mar  2 10:43:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 12:43:13 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
Message-ID: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>

I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
none of them are Centos 5.3
if you can reproduce it on a box I can visit (Centos 5.3 would be nice), put
in my ssh key and i'll come and look.

Be sure to remember, as you continue to add to this thread that you are
complaining about FS using 2% of your cpu......
That's somewhat amusing to me.


On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Sergey,
>
> that's not a problem, is how FS works.
>
> That load on the CPU when idle is actually not important. Is just the
> minimum, like in a carburettor.
>
> If you put it under load, you will see that there will need some
> serious load (many calls) to have it ramp up the cpu usage.
>
> -giovanni
>
> On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
> > I see the same problem on 4 absolutely different boxes with different
> linux
> > kernels.
> >
> > On Tuesday 02 March 2010, Anthony Minessale wrote:
> >> Well, there you go.
> >> your box is having a hard time doing a simple 1ms sleep in a loop for
> the
> >> centralized core timer.
> >>
> >> It's either your kernel or your box.  My box and every box I have do not
> >> have that problem so you may want to consider a new one.
> >>
> >> FreeSWITCH is FREE so give it a good home.
> >>
> >> The good news is, you can make upwards of 1500 calls on that same 2% of
> cpu
> >> usage.
> >>
> >>
> >>
> >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >> > FS spends half of CPU here (main thread):
> >> >
> >> > #0  0xffffe424 in __kernel_vsyscall ()
> >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> >> > #2  0xb7e392eb in apr_sleep () from
> >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> >> >
> >> > The second half is spent here:
> >> >
> >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> >> > #3  0xb7e379a8 in dummy_worker () from
> >> > /opt/freeswitch/lib/libfreeswitch.so.1
> >> >
> >> > On Tuesday 02 March 2010, Moises Silva wrote:
> >> > > ps -LlFm -p `pidof freeswitch`
> >> > >
> >> > > That shows you CPU usage per FreeSWITCH thread
> >> > >
> >> > > Then
> >> > >
> >> > > pstack `pidof freeswitch`
> >> > >
> >> > > That shows the stack trace of every FreeSWITCH thread and then you
> can
> >> >
> >> > find
> >> >
> >> > > exactly what is doing.
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From freeswitch at cartissolutions.com  Tue Mar  2 10:48:34 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 02 Mar 2010 12:48:34 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021316.30842.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>	<201003021251.48760.sos@sokhapkin.dyndns.org>	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
Message-ID: <4B8D5D82.5020304@cartissolutions.com>

On 03/02/2010 12:16 PM, Sergey Okhapkin wrote:
> I see the same problem on 4 absolutely different boxes with different linux
> kernels.
>    

And why do you think this is a problem anyway?  If you want 0% cpu 
utilization, you can always power off the machine.  :-)


> On Tuesday 02 March 2010, Anthony Minessale wrote:
>    
>> Well, there you go.
>> your box is having a hard time doing a simple 1ms sleep in a loop for the
>> centralized core timer.
>>
>> It's either your kernel or your box.  My box and every box I have do not
>> have that problem so you may want to consider a new one.
>>
>> FreeSWITCH is FREE so give it a good home.
>>
>> The good news is, you can make upwards of 1500 calls on that same 2% of cpu
>> usage.
>>      
>
<snip snip>



From andrew at hijacked.us  Tue Mar  2 10:53:02 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 2 Mar 2010 13:53:02 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<201003021251.48760.sos@sokhapkin.dyndns.org>
	<191c3a031003021000n296bd128p185136185811303b@mail.gmail.com>
	<201003021316.30842.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
Message-ID: <20100302185302.GG1751@hijacked.us>

On Tue, Mar 02, 2010 at 12:43:13PM -0600, Anthony Minessale wrote:
> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> none of them are Centos 5.3
> if you can reproduce it on a box I can visit (Centos 5.3 would be nice), put
> in my ssh key and i'll come and look.
> 
> Be sure to remember, as you continue to add to this thread that you are
> complaining about FS using 2% of your cpu......
> That's somewhat amusing to me.
>

I can confirm this, I just checked 2 of my machines which run identical
hardware and the Ubuntu one has FS sitting about 7% cpu all the time
and the CentOS box has it sitting at 0% (I can't even see it on the
screen). Both machines are idle.

So the moral of the story is that the OS/hardware recommendations are
made for good reason. Disregard them at your own risk.

Andrew



From sos at sokhapkin.dyndns.org  Tue Mar  2 10:53:48 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 2 Mar 2010 13:53:48 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
Message-ID: <201003021353.48590.sos@sokhapkin.dyndns.org>

A program should not be visible in "top" when idle. That's my point.

On Tuesday 02 March 2010, Anthony Minessale wrote:
> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> none of them are Centos 5.3
> if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
> put in my ssh key and i'll come and look.
>
> Be sure to remember, as you continue to add to this thread that you are
> complaining about FS using 2% of your cpu......
> That's somewhat amusing to me.
>
> On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli 
<gmaruzz at celliax.org>wrote:
> > Sergey,
> >
> > that's not a problem, is how FS works.
> >
> > That load on the CPU when idle is actually not important. Is just the
> > minimum, like in a carburettor.
> >
> > If you put it under load, you will see that there will need some
> > serious load (many calls) to have it ramp up the cpu usage.
> >
> > -giovanni
> >
> > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org> wrote:
> > > I see the same problem on 4 absolutely different boxes with different
> >
> > linux
> >
> > > kernels.
> > >
> > > On Tuesday 02 March 2010, Anthony Minessale wrote:
> > >> Well, there you go.
> > >> your box is having a hard time doing a simple 1ms sleep in a loop for
> >
> > the
> >
> > >> centralized core timer.
> > >>
> > >> It's either your kernel or your box.  My box and every box I have do
> > >> not have that problem so you may want to consider a new one.
> > >>
> > >> FreeSWITCH is FREE so give it a good home.
> > >>
> > >> The good news is, you can make upwards of 1500 calls on that same 2%
> > >> of
> >
> > cpu
> >
> > >> usage.
> > >>
> > >>
> > >>
> > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> > >>
> > >> <sos at sokhapkin.dyndns.org>wrote:
> > >> > FS spends half of CPU here (main thread):
> > >> >
> > >> > #0  0xffffe424 in __kernel_vsyscall ()
> > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > >> > #2  0xb7e392eb in apr_sleep () from
> > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> > >> >
> > >> > The second half is spent here:
> > >> >
> > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > >> > #3  0xb7e379a8 in dummy_worker () from
> > >> > /opt/freeswitch/lib/libfreeswitch.so.1
> > >> >
> > >> > On Tuesday 02 March 2010, Moises Silva wrote:
> > >> > > ps -LlFm -p `pidof freeswitch`
> > >> > >
> > >> > > That shows you CPU usage per FreeSWITCH thread
> > >> > >
> > >> > > Then
> > >> > >
> > >> > > pstack `pidof freeswitch`
> > >> > >
> > >> > > That shows the stack trace of every FreeSWITCH thread and then you
> >
> > can
> >
> > >> > find
> > >> >
> > >> > > exactly what is doing.
> > >> >
> > >> > _______________________________________________
> > >> > FreeSWITCH-users mailing list
> > >> > FreeSWITCH-users at lists.freeswitch.org
> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > >> > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From anthony.minessale at gmail.com  Tue Mar  2 11:05:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 13:05:06 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021353.48590.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <191c3a031003021105g51d73cfbs7b88562586366c28@mail.gmail.com>

FS is not really idle when idle.  Calls are not the only thing that cost
cpu.

Its a server daemon.  It has at least 25 threads up including the central
timer thread which is doing an endless-loop of 1ms sleeps.  You have an
active sip engine that still runs in a loop and many other moving parts.







On Tue, Mar 2, 2010 at 12:53 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> A program should not be visible in "top" when idle. That's my point.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
> > I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
> > none of them are Centos 5.3
> > if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
> > put in my ssh key and i'll come and look.
> >
> > Be sure to remember, as you continue to add to this thread that you are
> > complaining about FS using 2% of your cpu......
> > That's somewhat amusing to me.
> >
> > On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
> > > Sergey,
> > >
> > > that's not a problem, is how FS works.
> > >
> > > That load on the CPU when idle is actually not important. Is just the
> > > minimum, like in a carburettor.
> > >
> > > If you put it under load, you will see that there will need some
> > > serious load (many calls) to have it ramp up the cpu usage.
> > >
> > > -giovanni
> > >
> > > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org> wrote:
> > > > I see the same problem on 4 absolutely different boxes with different
> > >
> > > linux
> > >
> > > > kernels.
> > > >
> > > > On Tuesday 02 March 2010, Anthony Minessale wrote:
> > > >> Well, there you go.
> > > >> your box is having a hard time doing a simple 1ms sleep in a loop
> for
> > >
> > > the
> > >
> > > >> centralized core timer.
> > > >>
> > > >> It's either your kernel or your box.  My box and every box I have do
> > > >> not have that problem so you may want to consider a new one.
> > > >>
> > > >> FreeSWITCH is FREE so give it a good home.
> > > >>
> > > >> The good news is, you can make upwards of 1500 calls on that same 2%
> > > >> of
> > >
> > > cpu
> > >
> > > >> usage.
> > > >>
> > > >>
> > > >>
> > > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
> > > >>
> > > >> <sos at sokhapkin.dyndns.org>wrote:
> > > >> > FS spends half of CPU here (main thread):
> > > >> >
> > > >> > #0  0xffffe424 in __kernel_vsyscall ()
> > > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
> > > >> > #2  0xb7e392eb in apr_sleep () from
> > > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
> > > >> >
> > > >> > The second half is spent here:
> > > >> >
> > > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
> > > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
> > > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
> > > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
> > > >> > #3  0xb7e379a8 in dummy_worker () from
> > > >> > /opt/freeswitch/lib/libfreeswitch.so.1
> > > >> >
> > > >> > On Tuesday 02 March 2010, Moises Silva wrote:
> > > >> > > ps -LlFm -p `pidof freeswitch`
> > > >> > >
> > > >> > > That shows you CPU usage per FreeSWITCH thread
> > > >> > >
> > > >> > > Then
> > > >> > >
> > > >> > > pstack `pidof freeswitch`
> > > >> > >
> > > >> > > That shows the stack trace of every FreeSWITCH thread and then
> you
> > >
> > > can
> > >
> > > >> > find
> > > >> >
> > > >> > > exactly what is doing.
> > > >> >
> > > >> > _______________________________________________
> > > >> > FreeSWITCH-users mailing list
> > > >> > FreeSWITCH-users at lists.freeswitch.org
> > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > >> > http://www.freeswitch.org
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >s http://www.freeswitch.org
> > >
> > > --
> > > Sincerely,
> > >
> > > Giovanni Maruzzelli
> > > Cell : +39-347-2665618
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  2 11:05:47 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 14:05:47 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com><191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <1579D06951D94A4DAF34949175410B1F@MOBILEE1705>


----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 02, 2010 1:53 PM
Subject: Re: [Freeswitch-users] What FS is doing when idle?


>A program should not be visible in "top" when idle. That's my point.
>
> On Tuesday 02 March 2010, Anthony Minessale wrote:
>> I guess if you insist, you have 4 bad boxes with 4 bad kernels, and I bet
>> none of them are Centos 5.3
>> if you can reproduce it on a box I can visit (Centos 5.3 would be nice),
>> put in my ssh key and i'll come and look.
>>
>> Be sure to remember, as you continue to add to this thread that you are
>> complaining about FS using 2% of your cpu......
>> That's somewhat amusing to me.
>>
>> On Tue, Mar 2, 2010 at 12:23 PM, Giovanni Maruzzelli
> <gmaruzz at celliax.org>wrote:
>> > Sergey,
>> >
>> > that's not a problem, is how FS works.
>> >
>> > That load on the CPU when idle is actually not important. Is just the
>> > minimum, like in a carburettor.
>> >
>> > If you put it under load, you will see that there will need some
>> > serious load (many calls) to have it ramp up the cpu usage.
>> >
>> > -giovanni
>> >
>> > On Tue, Mar 2, 2010 at 7:16 PM, Sergey Okhapkin
>> >
>> > <sos at sokhapkin.dyndns.org> wrote:
>> > > I see the same problem on 4 absolutely different boxes with different
>> >
>> > linux
>> >
>> > > kernels.
>> > >
>> > > On Tuesday 02 March 2010, Anthony Minessale wrote:
>> > >> Well, there you go.
>> > >> your box is having a hard time doing a simple 1ms sleep in a loop 
>> > >> for
>> >
>> > the
>> >
>> > >> centralized core timer.
>> > >>
>> > >> It's either your kernel or your box.  My box and every box I have do
>> > >> not have that problem so you may want to consider a new one.
>> > >>
>> > >> FreeSWITCH is FREE so give it a good home.
>> > >>
>> > >> The good news is, you can make upwards of 1500 calls on that same 2%
>> > >> of
>> >
>> > cpu
>> >
>> > >> usage.
>> > >>
>> > >>
>> > >>
>> > >> On Tue, Mar 2, 2010 at 11:51 AM, Sergey Okhapkin
>> > >>
>> > >> <sos at sokhapkin.dyndns.org>wrote:
>> > >> > FS spends half of CPU here (main thread):
>> > >> >
>> > >> > #0  0xffffe424 in __kernel_vsyscall ()
>> > >> > #1  0xb7a82781 in select () from /lib/libc.so.6
>> > >> > #2  0xb7e392eb in apr_sleep () from
>> > >> > /opt/freeswitch/lib/libfreeswitch.so.1 #3  0x00000000 in ?? ()
>> > >> >
>> > >> > The second half is spent here:
>> > >> >
>> > >> > #0  0xb7d50946 in clock_nanosleep () from /lib/librt.so.1
>> > >> > #1  0xb7e0fa7d in softtimer_runtime () at src/switch_time.c:155
>> > >> > #2  0xb7dc585a in switch_loadable_module_exec (thread=0xb7101fb8,
>> > >> > obj=0xb7101da8) at src/switch_loadable_module.c:96
>> > >> > #3  0xb7e379a8 in dummy_worker () from
>> > >> > /opt/freeswitch/lib/libfreeswitch.so.1
>> > >> >
>> > >> > On Tuesday 02 March 2010, Moises Silva wrote:
>> > >> > > ps -LlFm -p `pidof freeswitch`
>> > >> > >
>> > >> > > That shows you CPU usage per FreeSWITCH thread
>> > >> > >
>> > >> > > Then
>> > >> > >
>> > >> > > pstack `pidof freeswitch`
>> > >> > >
>> > >> > > That shows the stack trace of every FreeSWITCH thread and then 
>> > >> > > you
>> >
>> > can
>> >
>> > >> > find
>> > >> >
>> > >> > > exactly what is doing.
>> > >> >
>> > >> > _______________________________________________
>> > >> > FreeSWITCH-users mailing list
>> > >> > FreeSWITCH-users at lists.freeswitch.org
>> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >> > UNSUBSCRIBE:
>> >
>> > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >
>> > >> > http://www.freeswitch.org
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
>> > >s http://www.freeswitch.org
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



I follow this thread, and did top on my Linux Fedora 10 64bits and 
freeswitch last svn version (without mod_vmd that not compiles anymore)
in idle mode and I get

10399 root      20   0  438m  37m 8744 S 11.3  0.5  15:31.61 freeswitch

if I remember the previous version took less resources

Regards

F 




From jonas.gauffin at gmail.com  Tue Mar  2 11:14:57 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 20:14:57 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
Message-ID: <d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>

Thank you! =)
The incoming INVITE had that header. I can't believe that I missed it in the
first place.

On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:

> Was the header there on the inbound invite?
>
> /b
>
> On Mar 2, 2010, at 12:11 PM, Jonas Gauffin wrote:
>
> Sure I can buy that, but not with a URI that doesnt match the inbound call.
>
> "0 <sip:0 at 212.151.Y.Y;user=phone>" fails number validation and therefore
> all calls through that javascript will be rejected by my provider.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Tue Mar  2 11:15:14 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Mar 2010 13:15:14 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021113.06820.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
Message-ID: <c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>

For whatever it's worth; I have FS running on a 1.6MHz processor,
CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.

On Tue, Mar 2, 2010 at 10:13 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> FS (r16858) consumes 4-5% of CPU time with spikes to 10% when idle. No single
> SIP request hits it. What does it doing? I unloaded all modules which can be
> unloaded, no difference. Tried with both -vm and -heavy-timer start up
> options. I'm running gentoo linux.
>
> # uname -a
> Linux ?2.6.28-tuxonice-r10 #1 SMP Sat Oct 17 08:58:15 EDT 2009 i686 Intel(R)
> Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Mar  2 11:25:29 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:25:29 -0600
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
	<d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
Message-ID: <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>

And are you using Proxy Media?

/b

On Mar 2, 2010, at 1:14 PM, Jonas Gauffin wrote:

> Thank you! =)
> The incoming INVITE had that header. I can't believe that I missed it in the first place.
> 
> On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:
> Was the header there on the inbound invite?
> 
> /b

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From brian at freeswitch.org  Tue Mar  2 11:25:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:25:10 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
Message-ID: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>

How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

:P

/b

On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

> For whatever it's worth; I have FS running on a 1.6MHz processor,
> CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.




From lists at redbonez.net  Tue Mar  2 11:27:45 2010
From: lists at redbonez.net (Adam Ford)
Date: Tue, 2 Mar 2010 12:27:45 -0700
Subject: [Freeswitch-users] xml_cdr http post differentiation between legs
Message-ID: <034401caba3e$70bbb950$52332bf0$@net>

Is there any way to tell whether it is the a-leg or b-leg when receiving the
POST data from mod_xml_cdr?

 

I wrote a parser in Python which parses the a_*.cdr.xml log files, and only
grabs the info in the associated b-leg log files if it needs additional
information(for example - in the case of a call that ends with mod_fifo).  I
want to adapt this to using the POST features of mod_xml_cdr so that it can
be real time. 

 

Any suggestions are greatly appreciated,

-Adam 

 

 

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From brian at freeswitch.org  Tue Mar  2 11:30:36 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:30:36 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
Message-ID: <38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>

I need you to email me your username and password for this challenge so I can calc this to verify please.

/b

On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:

> You are both correct. 
> 
> Issue created.
> 
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
> Jonas,
> 	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.
> 
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
> 
> Thanks,
> Brian
> 
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
> 
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>> 
>> 
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
>> Hello,
>> 
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
>> 
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>> 
>> 
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>> 
>> Regards,
>>   Jonas
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar  2 11:41:06 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 13:41:06 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
Message-ID: <1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>

What really really concerns me about this is the fact this is MD5 digest auth.

Neither of these:
4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47

Are NOT the right length to be an MD5 of any auth data which would be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really does concern me.

Anyone else have input on this one?

/b



On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:

> You are both correct. 
> 
> Issue created.
> 
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
> Jonas,
> 	Also I need a complete pcap of this without modifications.  I can't hand calculate this to see what is going on without intact data. Please open a jira and email me the pcap in private if you must.
> 
> Remember the list is not a bug tracker... Thats what we have jira for  ;)
> 
> Thanks,
> Brian
> 
> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
> 
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>> 
>> 
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
>> Hello,
>> 
>> I got a problem with rejected calls due to 403 (happens sometimes).
>> My sip provider found the problem, the nonce used by FS is sometimes incorrect (I got the trace from my sip provider).
>> 
>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>> 
>> 
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> EXECUTE sofia/external/061121487 at 212.151.144.8 bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227 (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83 sofia/external/0707728XXX SOFIA INIT
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111 (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338 (sofia/external/0707728XXX) State INIT going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132 sofia/external/0707728XXX SOFIA ROUTING
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66 (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/0707728XXX [BREAK]
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341 (sofia/external/0707728XXX) State ROUTING going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314 (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA
>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360 (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][0]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [calling][100]
>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel sofia/external/0707728XXX entering state [terminated][407]
>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>> 
>> Regards,
>>   Jonas
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From chris.chen2004 at gmail.com  Tue Mar  2 12:08:11 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 2 Mar 2010 15:08:11 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
Message-ID: <507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>

I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is
common to net books from ASUS and like, I have one running on EEE box N270.

Chris

On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> How many calls can that 1.6MHz CPU do?  :P And how long does it take to
> compile?
>
> :P
>
> /b
>
> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>
> > For whatever it's worth; I have FS running on a 1.6MHz processor,
> > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Tue Mar  2 12:09:56 2010
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 2 Mar 2010 14:09:56 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
Message-ID: <c55d46951003021209w45c35a9dw57d97f219236d09@mail.gmail.com>

Next time i compile I will let you know. But if I remember correctly,
it takes some few hours. And as far as calls are concerned I don't
intend  to have more than 8 calls at a time. So the question is really
irrelevant for my settings ;)

Max.

On Tue, Mar 2, 2010 at 1:25 PM, Brian West <brian at freeswitch.org> wrote:
> How many calls can that 1.6MHz CPU do? ?:P And how long does it take to compile?
>
> :P
>
> /b
>
> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>
>> For whatever it's worth; I have FS running on a 1.6MHz processor,
>> CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From infos at madovsky.org  Tue Mar  2 12:19:31 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 15:19:31 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com><744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
Message-ID: <91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>

I tried to downgrade FS to version 16597, all compiles well (even mod_vmd)
so with exactly the same config as last svn trunk I get with -vm :

21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch -> CPU = 2.3%

and without -vm
25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch -> CPU 4.00%

which is less as the last version that takes almost 12%.

hope this helps

Regards

F





  ----- Original Message ----- 
  From: Chris Chen 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 3:08 PM
  Subject: Re: [Freeswitch-users] What FS is doing when idle?


  I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.

  Chris 


  On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

    How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

    :P

    /b


    On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

    > For whatever it's worth; I have FS running on a 1.6MHz processor,
    > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.



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





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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Tue Mar  2 12:22:23 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:22:23 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
Message-ID: <25008BD4-47DE-46BF-ABB9-777467E5E80C@freeswitch.org>

I was just poking some fun... I knew he was on a 1.6ghz :P

/b

On Mar 2, 2010, at 2:08 PM, Chris Chen wrote:

> I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.
> 
> Chris 




From jonas.gauffin at gmail.com  Tue Mar  2 12:27:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:27:54 +0100
Subject: [Freeswitch-users] Diversion header
In-Reply-To: <1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>
References: <d021d3be1003020820q2bb4dbb8odffa309f94cd99fe@mail.gmail.com>
	<d021d3be1003020935p71050577k4f610fb4db83ed5a@mail.gmail.com>
	<6CAB4A3B-FEA8-467E-B7DB-A33862E082A1@freeswitch.org>
	<d021d3be1003021011o5221f75cu1d4f26c374d7da4d@mail.gmail.com>
	<9D3ECCF2-D1D1-4383-B171-3F185482A5A0@freeswitch.org>
	<d021d3be1003021114p69482909xcf0d0557e90df27d@mail.gmail.com>
	<1C400312-7E91-4572-95F6-DF20556EBE71@freeswitch.org>
Message-ID: <d021d3be1003021227r3eb9cbc6md5f337eba7d5ede@mail.gmail.com>

No. I'm only using proxy_media on fax devices.

On Tue, Mar 2, 2010 at 8:25 PM, Brian West <brian at freeswitch.org> wrote:

> And are you using Proxy Media?
>
> /b
>
> On Mar 2, 2010, at 1:14 PM, Jonas Gauffin wrote:
>
> Thank you! =)
> The incoming INVITE had that header. I can't believe that I missed it in
> the first place.
>
> On Tue, Mar 2, 2010 at 7:16 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Was the header there on the inbound invite?
>>
>> /b
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jonas.gauffin at gmail.com  Tue Mar  2 12:37:27 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:37:27 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
Message-ID: <d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>

Even if the string is incorrect, shouldn't FreeSWITCH use it without
modification in it's calculations (and return the same nonce in the
response)?

On Tue, Mar 2, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:

> What really really concerns me about this is the fact this is MD5 digest
> auth.
>
> Neither of these:
> 4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
> 4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47
>
> Are NOT the right length to be an MD5 of any auth data which would
> be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really
> does concern me.
>
> Anyone else have input on this one?
>
> /b
>
>
>
> On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:
>
> You are both correct.
>
> Issue created.
>
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Jonas,
>> Also I need a complete pcap of this without modifications.  I can't hand
>> calculate this to see what is going on without intact data. Please open a
>> jira and email me the pcap in private if you must.
>>
>> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>>
>> Thanks,
>> Brian
>>
>> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>>
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>>
>>
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I got a problem with rejected calls due to 403 (happens sometimes).
>>> My sip provider found the problem, the nonce used by FS is sometimes
>>> incorrect (I got the trace from my sip provider).
>>>
>>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>>
>>>
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>>> sofia/external/0707728XXX SOFIA INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>>> sofia/external/0707728XXX SOFIA ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [terminated][407]
>>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar  2 12:37:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 14:37:28 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com>
	<744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org>
	<507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com>
	<91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
Message-ID: <191c3a031003021237t30f46edp2978f258c54c326b@mail.gmail.com>

and make calls with both and measure the jitter with wireshark.


On Tue, Mar 2, 2010 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:

>  I tried to downgrade FS to version 16597, all compiles well (even
> mod_vmd)
> so with exactly the same config as last svn trunk I get with -vm :
>
> 21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch ->
> CPU = 2.3%
>
> and without -vm
> 25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch ->
> CPU 4.00%
>
> which is less as the last version that takes almost 12%.
>
> hope this helps
>
> Regards
>
> F
>
>
>
>
>  ----- Original Message -----
> *From:* Chris Chen <chris.chen2004 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 02, 2010 3:08 PM
> *Subject:* Re: [Freeswitch-users] What FS is doing when idle?
>
> I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is
> common to net books from ASUS and like, I have one running on EEE box N270.
>
> Chris
>
> On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:
>
>> How many calls can that 1.6MHz CPU do?  :P And how long does it take to
>> compile?
>>
>> :P
>>
>> /b
>>
>> On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:
>>
>> > For whatever it's worth; I have FS running on a 1.6MHz processor,
>> > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar  2 12:44:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:44:31 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<1A1F9843-6A36-4E3F-9BF8-77327B73DC1D@freeswitch.org>
	<d021d3be1003021237o1fd2fffi5e7f03ce5e66cb53@mail.gmail.com>
Message-ID: <F4B955FF-0B87-4CDC-92D4-04832A5004BB@freeswitch.org>

Not 100% sure since I don't have the details of your sip account to test this... I need to try this in eyebeam or something to see how it behaves.

/b

On Mar 2, 2010, at 2:37 PM, Jonas Gauffin wrote:

> Even if the string is incorrect, shouldn't FreeSWITCH use it without modification in it's calculations (and return the same nonce in the response)?
> 
> On Tue, Mar 2, 2010 at 8:41 PM, Brian West <brian at freeswitch.org> wrote:
> What really really concerns me about this is the fact this is MD5 digest auth.
> 
> Neither of these:
> 4b8cc1780000440a8d49b0854a8a7abb76fec907e6381eec
> 4b8cc1780000a33fc486f1017a5155a74fc6e3e611e28f47
> 
> Are NOT the right length to be an MD5 of any auth data which would be a33fc486f1017a5155a74fc6e3e611e28f47 so this stray 4b8cc1780000 really does concern me.
> 
> Anyone else have input on this one?
> 
> /b

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From msc at freeswitch.org  Tue Mar  2 12:44:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:44:34 -0800
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <201003021353.48590.sos@sokhapkin.dyndns.org>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
Message-ID: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>

On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> A program should not be visible in "top" when idle. That's my point.
>

That's a dangerous assumption. Do really know for a certainty that a program
"should not" be visible in top when it's "idle"? Like Tony said below,
"idle" is misleading. FreeSWITCH is working its but off when you think it's
"idle." Make 100 simultaneous calls and see what the CPU usage is. I think
you'll be happy with the results.

For the record, we are going to start a wiki page that explains all of this.
It's kind of analogous to the whole "Linux eats all my RAM" discussion. We
should have something documented in a few days.
-MC
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From msc at freeswitch.org  Tue Mar  2 12:48:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:48:30 -0800
Subject: [Freeswitch-users] wiki broken link pocketsphinx
In-Reply-To: <30970655E0CD476A95CF8382A2742084@MOBILEE1705>
References: <E5810B3B27964DDEAEE9B4A234BF57E2@MOBILEE1705>
	<5A9EAA2F-F7DA-4CC5-ACF6-AD9A0438964F@freeswitch.org>
	<30970655E0CD476A95CF8382A2742084@MOBILEE1705>
Message-ID: <87f2f3b91003021248s1f525a28xec3daefe20b9cd7c@mail.gmail.com>

On Tue, Mar 2, 2010 at 7:09 AM, Madovsky <infos at madovsky.org> wrote:

>  apparently Diego updated it.
> I don't know if I'm skilled enough to elaborate
>
>
AFAICT the link is correct now.
-MC
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From brian at freeswitch.org  Tue Mar  2 12:54:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:54:31 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <F58AF730-64DE-429A-A698-4DE13CC743FE@freeswitch.org>

Also add "Why is my load average so high with FreeSWITCH?" to that list of things we need to explain.  Because load average is just a metric of how many threads are in the run queue.. its really no way to tell how busy a box really is since most threads have narcolepsy.

/b

On Mar 2, 2010, at 2:44 PM, Michael Collins wrote:

> For the record, we are going to start a wiki page that explains all of this. It's kind of analogous to the whole "Linux eats all my RAM" discussion. We should have something documented in a few days.
> -MC




From jonas.gauffin at gmail.com  Tue Mar  2 12:55:28 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 2 Mar 2010 21:55:28 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
Message-ID: <d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>

Ok. I've done a digest calculation and got a match.
The "response" value is calculated using the nonce that FS sends in the
INVITE, and not the nonce that is recieved in the 407 reply from my
provider.


On Tue, Mar 2, 2010 at 8:30 PM, Brian West <brian at freeswitch.org> wrote:

> I need you to email me your username and password for this challenge so I
> can calc this to verify please.
>
> /b
>
> On Mar 2, 2010, at 11:04 AM, Jonas Gauffin wrote:
>
> You are both correct.
>
> Issue created.
>
> On Tue, Mar 2, 2010 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Jonas,
>> Also I need a complete pcap of this without modifications.  I can't hand
>> calculate this to see what is going on without intact data. Please open a
>> jira and email me the pcap in private if you must.
>>
>> Remember the list is not a bug tracker... Thats what we have jira for  ;)
>>
>> Thanks,
>> Brian
>>
>> On Mar 2, 2010, at 10:44 AM, Anthony Minessale wrote:
>>
>> Are you reporting what you think is a bug on the mailing list?
>> I think you have been around long enough to know better.............
>>
>>
>> On Tue, Mar 2, 2010 at 9:02 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I got a problem with rejected calls due to 403 (happens sometimes).
>>> My sip provider found the problem, the nonce used by FS is sometimes
>>> incorrect (I got the trace from my sip provider).
>>>
>>> A sip trace can be found here: http://pastebin.freeswitch.org/12280
>>>
>>>
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_core_session.c:1696 Execute
>>> bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> EXECUTE sofia/external/061121487 at 212.151.144.8bridge([origination_caller_id_number=02366153XX]sofia/gateway/tele2/0707728XXX)
>>> 2010-03-02 08:42:15.247995 [NOTICE] switch_channel.c:642 New Channel
>>> sofia/external/0707728XXX [085c1d55-f66e-4da3-9b75-325ab3d14b4d]
>>> 2010-03-02 08:42:15.247995 [DEBUG] mod_sofia.c:3227
>>> (sofia/external/0707728XXX) State Change CS_NEW -> CS_INIT
>>> 2010-03-02 08:42:15.247995 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:83
>>> sofia/external/0707728XXX SOFIA INIT
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:111
>>> (sofia/external/0707728XXX) State Change CS_INIT -> CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/external/0707728XXX) State INIT going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] mod_sofia.c:132
>>> sofia/external/0707728XXX SOFIA ROUTING
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_ivr_originate.c:66
>>> (sofia/external/0707728XXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_session.c:1012 Send signal
>>> sofia/external/0707728XXX [BREAK]
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:341
>>> (sofia/external/0707728XXX) State ROUTING going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/0707728XXX) Running State Change CS_CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA
>>> 2010-03-02 08:42:15.263619 [DEBUG] switch_core_state_machine.c:360
>>> (sofia/external/0707728XXX) State CONSUME_MEDIA going to sleep
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.263619 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][0]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.294868 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [calling][100]
>>> 2010-03-02 08:42:15.310492 [DEBUG] sofia.c:3992 Channel
>>> sofia/external/0707728XXX entering state [terminated][407]
>>> 2010-03-02 08:42:15.310492 [NOTICE] sofia.c:4634 Hangup
>>> sofia/external/0707728XXX [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>>
>>> Regards,
>>>   Jonas
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar  2 12:55:38 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 12:55:38 -0800
Subject: [Freeswitch-users] NAT issue?
In-Reply-To: <4B8D2BA6.1030907@cachecomm.com>
References: <4B8BEE7F.1030401@cachecomm.com>
	<87f2f3b91003010924p16400e10w67e3e90802f78075@mail.gmail.com>
	<4B8C273C.6010808@cachecomm.com>
	<FE7B4827-EC4D-4A71-AA8C-33E673E81468@freeswitch.org>
	<65d96fc81003011537kb5fd4c7xdb5df0b83fba3482@mail.gmail.com>
	<4B8C5AAE.2030908@cachecomm.com>
	<45C9B1E4-AE67-4E20-A5AB-71EB46EA0CC9@freeswitch.org>
	<65d96fc81003020103x7df499ecu3d9add7f3053a459@mail.gmail.com>
	<FE789B7F-8ED2-42A0-86CF-A3571348BEC1@freeswitch.org>
	<4B8D2BA6.1030907@cachecomm.com>
Message-ID: <87f2f3b91003021255m1b27caf3gc3ef95645e4162ab@mail.gmail.com>

I removed the Netgear P.O.S . and put the phones behind a linux firewall all
is good.

Let's put this text in neon letters ten feet high!
-MC
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From brian at freeswitch.org  Tue Mar  2 12:59:10 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 14:59:10 -0600
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<191c3a031003020844h44f63842pb4c88f4ebee088ec@mail.gmail.com>
	<151E6027-B371-4DA0-887C-8A8BAC336709@freeswitch.org>
	<d021d3be1003020904j4a39b967l658ac7561cfe5745@mail.gmail.com>
	<38BAC35D-F9B5-4133-AD78-5D833C3107FC@freeswitch.org>
	<d021d3be1003021255od422dd5p1ee32d34014d5405@mail.gmail.com>
Message-ID: <1C428CC2-8166-4061-9378-9FEC0388EDB6@freeswitch.org>

Please include all these details in the jira.

/b

On Mar 2, 2010, at 2:55 PM, Jonas Gauffin wrote:

> Ok. I've done a digest calculation and got a match.
> The "response" value is calculated using the nonce that FS sends in the INVITE, and not the nonce that is recieved in the 407 reply from my provider.
> 
> 
> On Tue, Mar 2, 2010 at 8:30 PM, Brian West <brian at freeswitch.org> wrote:
> I need you to email me your username and password for this challenge so I can calc this to verify please.
> 
> /b

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From anthony.minessale at gmail.com  Tue Mar  2 13:05:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Mar 2010 15:05:04 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <191c3a031003021305k7de4b3f8ie6bebebd19ceb8a3@mail.gmail.com>

Since this topic is philosophical to begin with, "What does a FS box do when
idle, and if a tree fell in the woods while a FS box was in a nearby cabin
would a bear shit in the woods....."

Another philosophical question would be, If a mailing-list thread persisted
for more than 6 hours would it take more or less time to just change the
code to make people stop complaining about free software  than it does to
politely not concern one's self with the issue.  I chose the first option,
you can update to trunk now and probably see your precious 0.0% so you can
get absolutely no use from your computer.  As a side note, my change
effectively proves  that the performance of pthread's mutexing and
conditional broadcast implementation has gone downhill since the version on
CentOS.


On Tue, Mar 2, 2010 at 2:44 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> A program should not be visible in "top" when idle. That's my point.
>>
>
> That's a dangerous assumption. Do really know for a certainty that a
> program "should not" be visible in top when it's "idle"? Like Tony said
> below, "idle" is misleading. FreeSWITCH is working its but off when you
> think it's "idle." Make 100 simultaneous calls and see what the CPU usage
> is. I think you'll be happy with the results.
>
> For the record, we are going to start a wiki page that explains all of
> this. It's kind of analogous to the whole "Linux eats all my RAM"
> discussion. We should have something documented in a few days.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From phunk0000 at hotmail.com  Tue Mar  2 13:15:06 2010
From: phunk0000 at hotmail.com (Todd)
Date: Tue, 2 Mar 2010 16:15:06 -0500
Subject: [Freeswitch-users] Play message upon call anwser
Message-ID: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks 

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From freeswitch at cartissolutions.com  Tue Mar  2 13:27:50 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Tue, 02 Mar 2010 15:27:50 -0600
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
Message-ID: <4B8D82D6.90007@cartissolutions.com>

On 03/02/2010 02:44 PM, Michael Collins wrote:
>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin 
> <sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>> wrote:
>
>     A program should not be visible in "top" when idle. That's my point.
>
>
> That's a dangerous assumption. Do really know for a certainty that a 
> program "should not" be visible in top when it's "idle"? Like Tony 
> said below, "idle" is misleading. FreeSWITCH is working its but off 
> when you think it's "idle." Make 100 simultaneous calls and see what 
> the CPU usage is. I think you'll be happy with the results.
>
<snip snip>

Cue up applicable car analogy....

I ask Sergey this question:  When you turn on your car and let it while 
in either a parking gear or in neutral, your car is considered 
"idling".  But does the car still not use oxygen and gasoline, and 
produce heat and waste products all the while idling?  That is how you 
can think of a complex system such as FreeSWITCH.  It is a machine with 
a lot of parts, and idling means that it is in a "ready" state, but does 
not mean that it is "turned off".

-Yossi

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From robert.hadley at teotech.com  Tue Mar  2 13:31:24 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Tue, 2 Mar 2010 13:31:24 -0800
Subject: [Freeswitch-users] Delete custom voicemail greetings
Message-ID: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>

Hi FS users,

 

How do I delete custom voicemail greetings and go back to the default
greeting? 

 

Thanks,

Robert

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From pjintheusa at gmail.com  Tue Mar  2 13:40:30 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 2 Mar 2010 16:40:30 -0500
Subject: [Freeswitch-users] .NET / ESL
In-Reply-To: <251258.58331.qm@web33508.mail.mud.yahoo.com>
References: <367751821003011901p55388bd6h9f543fed4c3de69c@mail.gmail.com>
	<251258.58331.qm@web33508.mail.mud.yahoo.com>
Message-ID: <367751821003021340m4708598bu6c142fd74a0757d@mail.gmail.com>

Thank you Diego - I will be taking a look later this week and will let you
know how it goes.

Or see you on the conf call tomorrow!

On Tue, Mar 2, 2010 at 8:24 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi Phillip Jones,
>
> Any concerns, You tell me, I help you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
> --- On Mon, 3/1/10, Phillip Jones <pjintheusa at gmail.com> wrote:
>
> > From: Phillip Jones <pjintheusa at gmail.com>
> > Subject: Re: [Freeswitch-users] .NET / ESL
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Monday, March 1, 2010, 10:01 PM
> > Thanks
> >
> > On Mon, Mar 1, 2010 at 9:42 PM,
> > Brian West <brian at freeswitch.org>
> > wrote:
> >
> > http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/
> >
> >
> >
> > in the managed folder.
> >
> >
> >
> > /b
> >
> >
> >
> > On Mar 1, 2010, at 8:35 PM, Phillip Jones wrote:
> >
> >
> >
> > > Hi there,
> >
> > >
> >
> > > I am sure that I have read somewhere that someone is
> > developing a new .NET library for FreeSWITCH ESL. If so can
> > someone point me to some info / code etc. so I can take a
> > look.
> >
> > >
> >
> > > Thanks
> >
> > >
> >
> > > pj
> >
> > >
> > _______________________________________________
> >
> > > FreeSWITCH-users mailing list
> >
> > > FreeSWITCH-users at lists.freeswitch.org
> >
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > http://www.freeswitch.org
> >
> >
> >
> >
> >
> > _______________________________________________
> >
> > FreeSWITCH-users mailing list
> >
> > FreeSWITCH-users at lists.freeswitch.org
> >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > http://www.freeswitch.org
> >
> >
> >
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar  2 17:01:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 17:01:41 -0800
Subject: [Freeswitch-users] What FS is doing when idle?
In-Reply-To: <4B8D82D6.90007@cartissolutions.com>
References: <201003021113.06820.sos@sokhapkin.dyndns.org>
	<7b197bef1003021023o7ec6fba6sccc9b4185640a32a@mail.gmail.com>
	<191c3a031003021043o1d7a10aar3e5f88dad2738815@mail.gmail.com>
	<201003021353.48590.sos@sokhapkin.dyndns.org>
	<87f2f3b91003021244q3e9e4a90m5116fe4e77cd2e3a@mail.gmail.com>
	<4B8D82D6.90007@cartissolutions.com>
Message-ID: <87f2f3b91003021701i6848c9e9hfb665ee0f1892802@mail.gmail.com>

It's sad that my computer uses more than 2% CPU whenever I try to open this
thread in my email client...
-MC

On Tue, Mar 2, 2010 at 1:27 PM, Yossi Neiman <freeswitch at cartissolutions.com
> wrote:

>  On 03/02/2010 02:44 PM, Michael Collins wrote:
>
>
>
> On Tue, Mar 2, 2010 at 10:53 AM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> A program should not be visible in "top" when idle. That's my point.
>>
>
> That's a dangerous assumption. Do really know for a certainty that a
> program "should not" be visible in top when it's "idle"? Like Tony said
> below, "idle" is misleading. FreeSWITCH is working its but off when you
> think it's "idle." Make 100 simultaneous calls and see what the CPU usage
> is. I think you'll be happy with the results.
>
>   <snip snip>
>
> Cue up applicable car analogy....
>
> I ask Sergey this question:  When you turn on your car and let it while in
> either a parking gear or in neutral, your car is considered "idling".  But
> does the car still not use oxygen and gasoline, and produce heat and waste
> products all the while idling?  That is how you can think of a complex
> system such as FreeSWITCH.  It is a machine with a lot of parts, and idling
> means that it is in a "ready" state, but does not mean that it is "turned
> off".
>
> -Yossi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Mar  2 17:26:52 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 09:26:52 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
Message-ID: <23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>

Thanks, I removed @30i and it works, however, CELT, doesn't.

CODECS IN               PCMU,PCMA,iLBC,CELT,GSM
CODECS OUT              PCMU,PCMA,iLBC,CELT,GSM


2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:48000:10]/[GSM:3:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[GSM:3:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:24:58.190599 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[GSM:3:8000:20]

CELT at 32000h

2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CELT:114:32000:10]/[GSM:3:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
payload to 101
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [telephone-event:101:8000:10]/[GSM:3:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMU:0:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[PCMA:8:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[iLBC:98:8000:20]
2010-03-02 20:26:02.439565 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [CN:13:8000:10]/[GSM:3:8000:20]

2010/3/2 Brian West <brian at freeswitch.org>:
> Yes that is because its @30i, All codecs offered MUST be 30ms if you want to offer iLBC at 30i. ?So add @30i to all but GSM since you can't do GSM @ 30.
>
> /b
> On Mar 2, 2010, at 1:01 AM, Seven Du wrote:
>
>> sorry forget to metion I'm on FreeSWITCH Version 1.0.trunk (16859)
>>
>>
>> 2010/3/2 Seven Du <dujinfang at gmail.com>:
>>> I have iLBC configured in profile internal
>>>
>>> CODECS IN ? ? ? ? ? ? ? PCMU,PCMA,iLBC at 30i,GSM
>>> CODECS OUT ? ? ? ? ? ? ?PCMU,PCMA,iLBC at 30i,GSM
>>>
>>> and ? ? <param name="inbound-codec-negotiation" value="generous"/>
>>>
>>> but it seems not comparing with iLBC
>>>
>>> ?010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3489 Set 2833 dtmf
>>> payload to 101
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [telephone-event:101:8000:30]/[GSM:3:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[PCMU:0:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[PCMA:8:8000:20]
>>> 2010-03-02 01:43:48.183785 [DEBUG] sofia_glue.c:3533 Audio Codec
>>> Compare [CN:13:8000:30]/[GSM:3:8000:20]
>>>
>>>
>>> can anyone help on this? see full log at http://pastebin.freeswitch.org/12278 .
>>>
>>> Thanks.
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Mar  2 17:45:30 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 19:45:30 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable  destination
In-Reply-To: <23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
Message-ID: <AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>

Celt can only do 10ms :P

/b

On Mar 2, 2010, at 7:26 PM, Seven Du wrote:

> Thanks, I removed @30i and it works, however, CELT, doesn't.




From infos at madovsky.org  Tue Mar  2 18:47:25 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 21:47:25 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>
Message-ID: <7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>

make install samples
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 4:31 PM
  Subject: [Freeswitch-users] Delete custom voicemail greetings


  Hi FS users,

   

  How do I delete custom voicemail greetings and go back to the default greeting? 

   

  Thanks,

  Robert



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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue Mar  2 18:56:57 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Mar 2010 21:56:57 -0500
Subject: [Freeswitch-users] What FS is doing when idle?
References: <201003021113.06820.sos@sokhapkin.dyndns.org><c55d46951003021115v2f882ae2q5e62d04dfa950cc6@mail.gmail.com><744EE96D-9C50-486E-A19E-ADD353882C46@freeswitch.org><507898381003021208w1e1e14bey27e959a61060879@mail.gmail.com><91CC3A53929A419FB0E103BC0BEF95F1@MOBILEE1705>
	<191c3a031003021237t30f46edp2978f258c54c326b@mail.gmail.com>
Message-ID: <509D15C949DC42898735CB1CC3CDD7E4@MOBILEE1705>

Ok I will do it, but I  can make only one call since I work alone on FS :D
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 02, 2010 3:37 PM
  Subject: Re: [Freeswitch-users] What FS is doing when idle?


  and make calls with both and measure the jitter with wireshark.



  On Tue, Mar 2, 2010 at 2:19 PM, Madovsky <infos at madovsky.org> wrote:

    I tried to downgrade FS to version 16597, all compiles well (even mod_vmd)
    so with exactly the same config as last svn trunk I get with -vm :

    21245 root      20   0  495m  29m 7428 S  2.3  0.4   0:00.71 freeswitch -> CPU = 2.3%

    and without -vm
    25744 root      20   0  431m  29m 7432 S  4.0  0.4   0:01.46 freeswitch -> CPU 4.00%

    which is less as the last version that takes almost 12%.

    hope this helps

    Regards

    F





      ----- Original Message ----- 
      From: Chris Chen 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 02, 2010 3:08 PM
      Subject: Re: [Freeswitch-users] What FS is doing when idle?


      I guess he is thinking about 1.6GHZ, most likely the N270 CPU which is common to net books from ASUS and like, I have one running on EEE box N270.

      Chris 


      On Tue, Mar 2, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

        How many calls can that 1.6MHz CPU do?  :P And how long does it take to compile?

        :P

        /b


        On Mar 2, 2010, at 1:15 PM, Max Bridgewater wrote:

        > For whatever it's worth; I have FS running on a 1.6MHz processor,
        > CentOS 5.3 (Final); when idle it consumes 0.3% of processor time.



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





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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dujinfang at gmail.com  Tue Mar  2 19:45:15 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 11:45:15 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
Message-ID: <23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>

Oh, yes, but then how I set the pref codec string? I want to also
support other codecs and other ms, now I set to

 PCMU,PCMA,iLBC,CELT,GSM

with the above string I can do 30ms iLBC


2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMU:0:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[PCMA:8:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[iLBC:98:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3533 Audio Codec
Compare [iLBC:97:8000:30]/[GSM:3:8000:20]
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:3580 Substituting
codec iLBC at 30i@8000h
2010-03-02 22:44:01.071059 [DEBUG] sofia_glue.c:2333 Set Codec
sofia/internal/1001 at 114.80.200.174 iLBC/8000 30 ms 240 samples


should I use another profile that only do CELT with 10ms?

2010/3/3 Brian West <brian at freeswitch.org>:
> Celt can only do 10ms :P
>
> /b
>
> On Mar 2, 2010, at 7:26 PM, Seven Du wrote:
>
>> Thanks, I removed @30i and it works, however, CELT, doesn't.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Mar  2 19:51:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Mar 2010 21:51:38 -0600
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable  destination
In-Reply-To: <23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
	<23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
Message-ID: <D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>

Enable late negotiation? And pick after the fact?

/b

On Mar 2, 2010, at 9:45 PM, Seven Du wrote:

> Oh, yes, but then how I set the pref codec string? I want to also
> support other codecs and other ms, now I set to
> 
> PCMU,PCMA,iLBC,CELT,GSM
> 
> with the above string I can do 30ms iLBC




From msc at freeswitch.org  Tue Mar  2 20:03:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Mar 2010 20:03:53 -0800
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
Message-ID: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

>  Hey List..  I am trying to setup my dial plan so whenever a call is
> answered a recording is played to the person who answered the phone before
> the call is connected?.  Any help would be great.   Thanks
>
Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC
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From dujinfang at gmail.com  Tue Mar  2 20:20:20 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 3 Mar 2010 12:20:20 +0800
Subject: [Freeswitch-users] help on codec negotiation,
	got incompatable 	destination
In-Reply-To: <D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>
References: <23f91031003012257s32ce6cf7vb6469933b7a4bd15@mail.gmail.com>
	<23f91031003012301y42bb9c04s80689a21c0eb7924@mail.gmail.com>
	<074F8BA1-718B-4E36-9CC1-0FA9D3579E1D@freeswitch.org>
	<23f91031003021726m3b19d685s2658c05a7868a80a@mail.gmail.com>
	<AE5E12AB-1E0D-4157-9C34-E4D43ECBFDAC@freeswitch.org>
	<23f91031003021945r4ab49d4fi5d230a8afac7fdf4@mail.gmail.com>
	<D1FE6DB0-EFE6-4047-A442-F34CEE2EACBA@freeswitch.org>
Message-ID: <23f91031003022020o24c809afs55afd9574afa4ee5@mail.gmail.com>

2010/3/3 Brian West <brian at freeswitch.org>:
> Enable late negotiation? And pick after the fact?

Understood. Is that the conf of freeswitch conference call?

But I think it would be simpler to start a new profile in my scenario.

And, doesn't it make sense to also compare mixed ms?


Thanks.
>
> /b
>
> On Mar 2, 2010, at 9:45 PM, Seven Du wrote:
>
>> Oh, yes, but then how I set the pref codec string? I want to also
>> support other codecs and other ms, now I set to
>>
>> PCMU,PCMA,iLBC,CELT,GSM
>>
>> with the above string I can do 30ms iLBC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From lakindia89 at gmail.com  Tue Mar  2 20:48:29 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Wed, 3 Mar 2010 10:18:29 +0530
Subject: [Freeswitch-users] make current - System Hangs
In-Reply-To: <2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>
References: <7d79b3931003012314r161c4120k960b34b3b98c533c@mail.gmail.com>
	<87f2f3b91003020940n4781a0a2j5c63c1d098b434e5@mail.gmail.com>
	<2C6C36D4-8709-42A1-9FD7-F9F7A39A7E6C@freeswitch.org>
Message-ID: <7d79b3931003022048y48c8ecbdrc1c0e72877a9f6c4@mail.gmail.com>

Hey! when I did the make current for several times yesterday, the system got
hanged, and on the same line only.

But today when I tried, it went fine. I've not done any changes other than
shutting the system and started it again..

This time I got revision 16873 with openap(1048).
But yesterday it was 16859 with openzap(1047) and I just have a though that
it may be because of this.
Thank u all.


On Tue, Mar 2, 2010 at 11:17 PM, Brian West <brian at freeswitch.org> wrote:

> what os are you on?
>
> /b
>
> On Mar 2, 2010, at 11:40 AM, Michael Collins wrote:
>
>
>
> On Mon, Mar 1, 2010 at 11:14 PM, lakshmanan ganapathy <
> lakindia89 at gmail.com> wrote:
>
>> Hi all,
>> I did make current from my freeswitch src directory, but the system got
>> hanged.
>> Any one know why this happened.
>>
>
> Does it happen every time you make current? If so, does it always happen at
> the "expat" lib?
> -MC
>
> _________
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Tue Mar  2 21:04:46 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 2 Mar 2010 23:04:46 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
Message-ID: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>

Hi all,

Is anyone running FreeSWITCH virtualized (xen or vmware) in  
production?  I saw ec2 and xen mentioned a few places.  If so, are  
there tips or tricks for handle timer issues which affect virtual  
machines in general, such as recommended divider or clocksource  
settings?  I'm seeing funny timing issues while testing with audio  
recording and playback that speed up or slow down.

I have tried both paravirt and hvm.  The default xen kernel has a  
timer of 250, so I tried to hvm instead using the stock 5.3 kernel  
with the 1khz timer but didn't see much improvement.

All of this was tested on 64-bit CentOS 5.3.
- Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1  
SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP  
Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan  
21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=1000


I'm sure physical hosts are the de-facto best practice, but are  
virtual machines not worth it at all?

Cheers,
Rob



From brian at microcomaustralia.com.au  Tue Mar  2 21:27:29 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 3 Mar 2010 16:27:29 +1100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
Message-ID: <3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>

On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production? ?I saw ec2 and xen mentioned a few places. ?If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings? ?I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.

I tried tdm400+xen DOM0+asterisk years ago, on a single processor
machine but gave up. zttest returned good results, but analogue
quality has intermittently very poor quality and it made telephone
conversations difficult to understand. Especially a problem when
hourly cron jobs would start on all VMs at exactly the same time, but
was a problem at other times too. Plus the fact people who used the
phone more often then me didn't report the problem, so I thought it
was fine.

Plus the general fact I didn't really like the idea of running
Asterisk on DOM0, and running it on DOMU seemed to imply even more
overheard, as DOM0 would have to process the interrupts and pass them
to the DOMU.

Unfortunately I don't remember what scheduler I used, maybe sedf?

I suspect a multi-processor system would be a lot better suited for
the task. Also might be OK if you don't use zaptel hardware... I seem
to remember these cards generate a huge number of interrupts.

While this was with Asterisk, I suspect my results were not specific
to Asterisk in anyway.
-- 
Brian May <brian at microcomaustralia.com.au>



From jmesquita at freeswitch.org  Tue Mar  2 22:00:11 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 3 Mar 2010 03:00:11 -0300
Subject: [Freeswitch-users] FScomm
In-Reply-To: <D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
	<D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
Message-ID: <ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>

This is definitely a problem with your Qt installation. You need to be able
to find the libs to link it, otherwise, unresolved symbols everywhere!

Regards,
Jo?o Mesquita
FSComm Developer


On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:

>
> ----- Original Message -----
> From: "Michael Jerris" <mike at jerris.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, February 24, 2010 3:40 AM
> Subject: Re: [Freeswitch-users] FScomm
>
>
>
> On Feb 23, 2010, at 6:18 PM, Madovsky wrote:
>
> >
> > ----- Original Message -----
> > From: Jeff Lenk
> > To: freeswitch-users at lists.freeswitch.org
> > Sent: Tuesday, February 23, 2010 5:05 PM
> > Subject: Re: [Freeswitch-users] FScomm
> >
> >
> > http://wiki.freeswitch.org/wiki/FSComm#Linux
> >
> > you must run those from the FSComm directory
> >
>
> ?.
>
> >
> > It's what I did,
> > but from FS trunk, inside fscomm directory,
> > there s only
> >
> > account.cpp  conf                fshost.h        mainwindow.ui
> > resources.qrc
> > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
> > call.cpp     FSComm.pro          mainwindow.cpp  preferences
> > call.h       fshost.cpp          mainwindow.h    resources
> >
>
>
> Read those installation instructions again and do them step by step, you
> skipped one.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Ok now I have
>
> [root at node250 fscomm]# qmake
> WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
> in SOURCES
> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
> HEADERS
> WARNING: Found potential symbol conflict of prefdialog.cpp
> (preferences/prefdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of prefdialog.h
> (preferences/prefdialog.h) in HEADERS
> WARNING: Found potential symbol conflict of accountdialog.cpp
> (preferences/accountdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of accountdialog.h
> (preferences/accountdialog.h) in HEADERS
>
> [root at node250 fscomm]# qmake
> WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
> in SOURCES
> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
> HEADERS
> WARNING: Found potential symbol conflict of prefdialog.cpp
> (preferences/prefdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of prefdialog.h
> (preferences/prefdialog.h) in HEADERS
> WARNING: Found potential symbol conflict of accountdialog.cpp
> (preferences/accountdialog.cpp) in SOURCES
> WARNING: Found potential symbol conflict of accountdialog.h
> (preferences/accountdialog.h) in HEADERS
> [root at node250 fscomm]# make
> Makefile:278: warning: overriding commands for target `prefdialog.o'
> Makefile:215: warning: ignoring old commands for target `prefdialog.o'
> Makefile:285: warning: overriding commands for target `accountdialog.o'
> Makefile:234: warning: ignoring old commands for target `accountdialog.o'
> Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
> Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
> Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
> Makefile:307: warning: ignoring old commands for target
> `moc_accountdialog.o'
> Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
> Makefile:326: warning: ignoring old commands for target
> `moc_mainwindow.cpp'
> Makefile:350: warning: overriding commands for target
> `preferences/moc_prefdialog.cpp'
> Makefile:332: warning: ignoring old commands for target
> `preferences/moc_prefdialog.cpp'
> Makefile:353: warning: overriding commands for target
> `preferences/moc_accountdialog.cpp'
> Makefile:341: warning: ignoring old commands for target
> `preferences/moc_accountdialog.cpp'
> g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
> -fexceptions
>  -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic
>  -DQT_NO_DEBUG
>  -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT
> -I/usr/lib64/qt-3.3/mkspecs/default
>  -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src
> -I/usr/lib64/qt-3.3/include
>  -o main.o main.cpp
> main.cpp:31:25: error: QSplashScreen: No such file or directory
> In file included from main.cpp:32:
> mainwindow.h:34:23: error: QMainWindow: No such file or directory
> mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
> mainwindow.h:36:25: error: QSignalMapper: No such file or directory
> mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
> In file included from mainwindow.h:39,
>                 from main.cpp:32:
> ./fshost.h:32:19: error: QThread: No such file or directory
> ./fshost.h:33:17: error: QHash: No such file or directory
> ./fshost.h:34:26: error: QSharedPointer: No such file or directory
> In file included from ./fshost.h:36,
>                 from mainwindow.h:39,
>                 from main.cpp:32:
> ./call.h:32:18: error: QtCore: No such file or directory
> ./call.h:33:19: error: QString: No such file or directory
> In file included from mainwindow.h:42,
>                 from main.cpp:32:
> preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
> preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
> directory
> preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
> In file included from ./fshost.h:37,
>                 from mainwindow.h:39,
>                 from main.cpp:32:
> ./account.h:18: error: expected constructor, destructor, or type conversion
> before ?static?
> In file included from mainwindow.h:39,
>                 from main.cpp:32:
> ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
> /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
> QThread?
> ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:46: error: expected ?;? before ?<? token
> ./fshost.h:47: error: expected `;' before ?QSharedPointer?
> ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:47: error: expected ?;? before ?<? token
> ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:48: error: template argument 1 is invalid
> ./fshost.h:48: error: expected unqualified-id before ?>? token
> ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:49: error: expected ?;? before ?<? token
> ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:50: error: expected ?;? before ?<? token
> ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
> no type
> ./fshost.h:51: error: expected ?;? before ?<? token
> ./fshost.h:52: error: ?QSharedPointer? has not been declared
> ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:60: error: ?QSharedPointer? has not been declared
> ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:61: error: ?QSharedPointer? has not been declared
> ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:62: error: ?QSharedPointer? has not been declared
> ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:63: error: ?QSharedPointer? has not been declared
> ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:64: error: ?QSharedPointer? has not been declared
> ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:65: error: ?QSharedPointer? has not been declared
> ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:66: error: ?QSharedPointer? has not been declared
> ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:67: error: ?QSharedPointer? has not been declared
> ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:71: error: ?QSharedPointer? has not been declared
> ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
> ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:78: error: template argument 2 is invalid
> ./fshost.h:78: error: expected unqualified-id before ?>? token
> ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
> ./fshost.h:79: error: template argument 2 is invalid
> ./fshost.h:79: error: expected unqualified-id before ?>? token
> ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
> In file included from mainwindow.h:42,
>                 from main.cpp:32:
> preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
> QDialog?
> /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
> QDialog?
> preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
> ?QSettings? with no type
> preferences/prefdialog.h:31: error: expected ?;? before ?*? token
> In file included from main.cpp:32:
> mainwindow.h:48: error: expected class-name before ?{? token
> mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
> mainwindow.h:71: error: ?QSharedPointer? has not been declared
> mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:72: error: ?QSharedPointer? has not been declared
> mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:73: error: ?QSharedPointer? has not been declared
> mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:74: error: ?QSharedPointer? has not been declared
> mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:75: error: ?QSharedPointer? has not been declared
> mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:78: error: ?QSharedPointer? has not been declared
> mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:79: error: ?QSharedPointer? has not been declared
> mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:80: error: ?QSharedPointer? has not been declared
> mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
> mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
> mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
> mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
> no type
> mainwindow.h:86: error: expected ?;? before ?*? token
> mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
> with no type
> mainwindow.h:88: error: expected ?;? before ?*? token
> main.cpp: In function ?int main(int, char**)?:
> main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
> type
> main.cpp:42: error: ?QSplashScreen? was not declared in this scope
> main.cpp:42: error: ?splash? was not declared in this scope
> main.cpp:42: error: expected type-specifier before ?QSplashScreen?
> main.cpp:42: error: expected `;' before ?QSplashScreen?
> main.cpp:48: error: no matching function for call to
> ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
> /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
> QObject::connect(const QObject*, const char*, const QObject*, const char*,
> Qt::ConnectionType)
> /usr/include/QtCore/qobject.h:307: note:                 bool
> QObject::connect(const QObject*, const char*, const char*,
> Qt::ConnectionType) const
> main.cpp:49: error: ?class FSHost? has no member named ?start?
> main.cpp:41: warning: unused variable ?image?
> ./fshost.h: At global scope:
> ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)?
> defined
> but not used
> make: *** [main.o] Error 1
>
>
> Any idea ?
>
> Thx
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Tue Mar  2 22:01:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 3 Mar 2010 03:01:46 -0300
Subject: [Freeswitch-users] How to exchange media when I developed new
	endpoint module?
In-Reply-To: <93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>
References: <93b0f8ce1002091914q581f12fvd1de01e2181f0f3a@mail.gmail.com>
	<ebbc2d551002091932u52dbf1b6jfecf2dfdb43b8a7c@mail.gmail.com>
	<4021C303-1AC3-48F2-A17B-63C02C1C54D4@freeswitch.org>
	<93b0f8ce1002191722h5782b787qe56a06dd0cb94e76@mail.gmail.com>
	<ebbc2d551002201327o7a7bad26g48cac48f610aed55@mail.gmail.com>
	<93b0f8ce1002201812x24cf3853u920bdba87f938b43@mail.gmail.com>
Message-ID: <ebbc2d551003022201p6a775347wbeea66c8d0e4c32e@mail.gmail.com>

I have added you on GTalk, let me know if you need help.

Regards,
Jo?o Mesquita


On Sat, Feb 20, 2010 at 11:12 PM, ??? <spiritonly at gmail.com> wrote:

> Oh, My God! You are the developer of mod_khomp! I read your blog, checked
> out mod_khomp from google code, and found it had not update for a long time.
> So what is your roadmap about mod_khomp?
>
> I share my snippet in FS pastebin, you can find it with '
> http://pastebin.freeswitch.org/12192 '.
>
> My MSN is spiritonly at live.cn.
> My Gtalk is spiritonly at gmail.com.
> Hope we can keep connected.
>
>
> 2010/2/21 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>  I developed the current implementation of mod_khomp. I wouldn't take it as
>> an example for anything since there has been no activity there for the past
>> 4 months. If you care to share a snippet of your code, maybe we can help
>> better.
>>
>>
>> JM
>>
>>
>>
>> On Fri, Feb 19, 2010 at 11:22 PM, ??? <spiritonly at gmail.com> wrote:
>>
>>> Do you know mod_khomp? You can found it in FS wiki. I am developing an
>>> endpoint module like it.
>>> So you can give me some advice to bridge two session?
>>>
>>>
>>> On Wed, Feb 10, 2010 at 11:44 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> But the bigger question is what protocol are you doing that you have to
>>>> create your own endpoint module?
>>>>
>>>> /b
>>>>
>>>> On Feb 9, 2010, at 9:32 PM, Jo?o Mesquita wrote:
>>>>
>>>> > You should look at read_frame and write_frame implementations of other
>>>> endpoint modules.
>>>> >
>>>> > This should pretty much tell you how things work...
>>>> >
>>>> > Jo?o Mesquita
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From infos at madovsky.org  Tue Mar  2 22:26:54 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 01:26:54 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705><COL117-W5563292C886257AFC56548B9420@phx.gbl><D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705><BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com><D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
	<ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>
Message-ID: <13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>

Hi Joao,

thanks for your answer.
I have a standard RPM QT last update for Fedora10 64bits.
so I really don't know what path FScomm needs

Regards

Franck
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 1:00 AM
  Subject: Re: [Freeswitch-users] FScomm


  This is definitely a problem with your Qt installation. You need to be able to find the libs to link it, otherwise, unresolved symbols everywhere!


  Regards,
  Jo?o Mesquita
  FSComm Developer



  On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:


    ----- Original Message -----
    From: "Michael Jerris" <mike at jerris.com>

    To: <freeswitch-users at lists.freeswitch.org>

    Sent: Wednesday, February 24, 2010 3:40 AM

    Subject: Re: [Freeswitch-users] FScomm



    On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

    >
    > ----- Original Message -----
    > From: Jeff Lenk
    > To: freeswitch-users at lists.freeswitch.org
    > Sent: Tuesday, February 23, 2010 5:05 PM
    > Subject: Re: [Freeswitch-users] FScomm
    >
    >
    > http://wiki.freeswitch.org/wiki/FSComm#Linux
    >
    > you must run those from the FSComm directory
    >

    ?.

    >
    > It's what I did,
    > but from FS trunk, inside fscomm directory,
    > there s only
    >
    > account.cpp  conf                fshost.h        mainwindow.ui
    > resources.qrc
    > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
    > call.cpp     FSComm.pro          mainwindow.cpp  preferences
    > call.h       fshost.cpp          mainwindow.h    resources
    >


    Read those installation instructions again and do them step by step, you
    skipped one.
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org



    Ok now I have

    [root at node250 fscomm]# qmake
    WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
    in SOURCES
    WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
    HEADERS
    WARNING: Found potential symbol conflict of prefdialog.cpp
    (preferences/prefdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of prefdialog.h
    (preferences/prefdialog.h) in HEADERS
    WARNING: Found potential symbol conflict of accountdialog.cpp
    (preferences/accountdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of accountdialog.h
    (preferences/accountdialog.h) in HEADERS

    [root at node250 fscomm]# qmake
    WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
    in SOURCES
    WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
    HEADERS
    WARNING: Found potential symbol conflict of prefdialog.cpp
    (preferences/prefdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of prefdialog.h
    (preferences/prefdialog.h) in HEADERS
    WARNING: Found potential symbol conflict of accountdialog.cpp
    (preferences/accountdialog.cpp) in SOURCES
    WARNING: Found potential symbol conflict of accountdialog.h
    (preferences/accountdialog.h) in HEADERS
    [root at node250 fscomm]# make
    Makefile:278: warning: overriding commands for target `prefdialog.o'
    Makefile:215: warning: ignoring old commands for target `prefdialog.o'
    Makefile:285: warning: overriding commands for target `accountdialog.o'
    Makefile:234: warning: ignoring old commands for target `accountdialog.o'
    Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
    Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
    Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
    Makefile:307: warning: ignoring old commands for target
    `moc_accountdialog.o'
    Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
    Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
    Makefile:350: warning: overriding commands for target
    `preferences/moc_prefdialog.cpp'
    Makefile:332: warning: ignoring old commands for target
    `preferences/moc_prefdialog.cpp'
    Makefile:353: warning: overriding commands for target
    `preferences/moc_accountdialog.cpp'
    Makefile:341: warning: ignoring old commands for target
    `preferences/moc_accountdialog.cpp'
    g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
     -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG
     -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default
     -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
     -o main.o main.cpp
    main.cpp:31:25: error: QSplashScreen: No such file or directory
    In file included from main.cpp:32:
    mainwindow.h:34:23: error: QMainWindow: No such file or directory
    mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
    mainwindow.h:36:25: error: QSignalMapper: No such file or directory
    mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
    In file included from mainwindow.h:39,
                    from main.cpp:32:
    ./fshost.h:32:19: error: QThread: No such file or directory
    ./fshost.h:33:17: error: QHash: No such file or directory
    ./fshost.h:34:26: error: QSharedPointer: No such file or directory
    In file included from ./fshost.h:36,
                    from mainwindow.h:39,
                    from main.cpp:32:
    ./call.h:32:18: error: QtCore: No such file or directory
    ./call.h:33:19: error: QString: No such file or directory
    In file included from mainwindow.h:42,
                    from main.cpp:32:
    preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
    preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
    directory
    preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
    In file included from ./fshost.h:37,
                    from mainwindow.h:39,
                    from main.cpp:32:
    ./account.h:18: error: expected constructor, destructor, or type conversion
    before ?static?
    In file included from mainwindow.h:39,
                    from main.cpp:32:
    ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
    /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
    QThread?
    ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:46: error: expected ?;? before ?<? token
    ./fshost.h:47: error: expected `;' before ?QSharedPointer?
    ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:47: error: expected ?;? before ?<? token
    ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:48: error: template argument 1 is invalid
    ./fshost.h:48: error: expected unqualified-id before ?>? token
    ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:49: error: expected ?;? before ?<? token
    ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:50: error: expected ?;? before ?<? token
    ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
    no type
    ./fshost.h:51: error: expected ?;? before ?<? token
    ./fshost.h:52: error: ?QSharedPointer? has not been declared
    ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:60: error: ?QSharedPointer? has not been declared
    ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:61: error: ?QSharedPointer? has not been declared
    ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:62: error: ?QSharedPointer? has not been declared
    ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:63: error: ?QSharedPointer? has not been declared
    ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:64: error: ?QSharedPointer? has not been declared
    ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:65: error: ?QSharedPointer? has not been declared
    ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:66: error: ?QSharedPointer? has not been declared
    ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:67: error: ?QSharedPointer? has not been declared
    ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:71: error: ?QSharedPointer? has not been declared
    ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
    ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:78: error: template argument 2 is invalid
    ./fshost.h:78: error: expected unqualified-id before ?>? token
    ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
    ./fshost.h:79: error: template argument 2 is invalid
    ./fshost.h:79: error: expected unqualified-id before ?>? token
    ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
    In file included from mainwindow.h:42,
                    from main.cpp:32:
    preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
    QDialog?
    /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
    QDialog?
    preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
    ?QSettings? with no type
    preferences/prefdialog.h:31: error: expected ?;? before ?*? token
    In file included from main.cpp:32:
    mainwindow.h:48: error: expected class-name before ?{? token
    mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
    mainwindow.h:71: error: ?QSharedPointer? has not been declared
    mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:72: error: ?QSharedPointer? has not been declared
    mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:73: error: ?QSharedPointer? has not been declared
    mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:74: error: ?QSharedPointer? has not been declared
    mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:75: error: ?QSharedPointer? has not been declared
    mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:78: error: ?QSharedPointer? has not been declared
    mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:79: error: ?QSharedPointer? has not been declared
    mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:80: error: ?QSharedPointer? has not been declared
    mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
    mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
    mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
    mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
    no type
    mainwindow.h:86: error: expected ?;? before ?*? token
    mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
    with no type
    mainwindow.h:88: error: expected ?;? before ?*? token
    main.cpp: In function ?int main(int, char**)?:
    main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
    type
    main.cpp:42: error: ?QSplashScreen? was not declared in this scope
    main.cpp:42: error: ?splash? was not declared in this scope
    main.cpp:42: error: expected type-specifier before ?QSplashScreen?
    main.cpp:42: error: expected `;' before ?QSplashScreen?
    main.cpp:48: error: no matching function for call to
    ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
    /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
    QObject::connect(const QObject*, const char*, const QObject*, const char*,
    Qt::ConnectionType)
    /usr/include/QtCore/qobject.h:307: note:                 bool
    QObject::connect(const QObject*, const char*, const char*,
    Qt::ConnectionType) const
    main.cpp:49: error: ?class FSHost? has no member named ?start?
    main.cpp:41: warning: unused variable ?image?
    ./fshost.h: At global scope:
    ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined
    but not used
    make: *** [main.o] Error 1


    Any idea ?

    Thx




    _______________________________________________
    FreeSWITCH-users mailing list
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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------------------------------------------------------------------------------


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From dome at tel.co.th  Tue Mar  2 22:38:58 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 3 Mar 2010 13:38:58 +0700
Subject: [Freeswitch-users] Nibblebill for NoSQL
Message-ID: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>

Dear All,
     I use nibllebill + postgresql for longtime. it' work wel when
concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
many problem and slow. so i have 2 choice.
1. upgrade hardware , add more memory may be 32 GB and change HDD to
SSD (or IOdrive ).
2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
I choose 2. and use tokyo tyrant. because it's persistant and support
memcache API.
  Today i create new module base on nibblebill but use memcacahe API
connect to tokyo tyrant. it's look fast. and work without any problem.
  If someone want to use my module I'l acttach to JIRA and create WIKI page.

Please let's me know if you want.

BG
Dome C.



From infos at madovsky.org  Tue Mar  2 22:49:34 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 01:49:34 -0500
Subject: [Freeswitch-users] Nibblebill for NoSQL
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <1B6F4F74BC724CCC8FB842F63D0279E0@MOBILEE1705>

Hi Dome,

postgresql is a very powerful DB that can accepts a lot of simultaneous 
connections
it needs to be set carefully to fit your memory and HD.
try google postgresql tuning
However the other solution can be interesting too.

Regards

F

----- Original Message ----- 
From: "Dome Charoenyost" <dome at tel.co.th>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 03, 2010 1:38 AM
Subject: [Freeswitch-users] Nibblebill for NoSQL


> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI 
> page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 




From mayamatakeshi at gmail.com  Wed Mar  3 00:06:28 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Wed, 3 Mar 2010 17:06:28 +0900
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
Message-ID: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>

Hello,
was any timing related change done recently in FS?
I have updated to trunk r. 16873 (was just some 200 revisions behind) and
now I'm seeing several messages like this when I try as little as 5 calls
per second:
2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle Error!
111

When this happens, FS refuses the related call with "503 Maximum Calls In
Progress" although, I'm using default config files (sessions-per-second=30).
So it seems to me FS is wrongly aggregating incoming calls on the same
interval.

I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
FS never showed this before in any of them.

regards,
takeshi
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From jonas.gauffin at gmail.com  Wed Mar  3 00:07:09 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 3 Mar 2010 09:07:09 +0100
Subject: [Freeswitch-users] OT: SIP and IP spoofing
Message-ID: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>

Hello,

My sip gateway provider are using both IP address locking (only my servers
IP addresses can use my gateway account) and Digest authentication on every
call.
I asked why and they said that the account would be vulnerable to IP
spoofing otherwise. Is that possible? I mean, if someone fakes my servers IP
address in the packets, shouldn't the responses be sent back to my server
and not the one creating the fake packets? Are there any other reasons to
use both ip locking and digest authentication?

Regards,
  Jonas
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From red.rain.seven at gmail.com  Wed Mar  3 00:07:42 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Wed, 3 Mar 2010 16:07:42 +0800
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <59ad9ca11003030007m262d1544hc72af6d2175c072e@mail.gmail.com>

Dome

I am very interested in your module. Please add it to JIRA and create a WIKI
page for it. And does everything else that used to be on your postgresql now
being migrated to the memcache API + tokyo tyrant?



On Wed, Mar 3, 2010 at 2:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jaybinks at gmail.com  Wed Mar  3 00:42:18 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 3 Mar 2010 18:42:18 +1000
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>

im VERY interested in this...
( and the use of NoSQL in general with FS )

please can you email over what you have, or post it to jira.

J

On Wed, Mar 3, 2010 at 4:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>     I use nibllebill + postgresql for longtime. it' work wel when
> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
> many problem and slow. so i have 2 choice.
> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
> SSD (or IOdrive ).
> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
> I choose 2. and use tokyo tyrant. because it's persistant and support
> memcache API.
>  Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>  If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From viper at fx-services.com  Wed Mar  3 05:49:54 2010
From: viper at fx-services.com (Robin Vleij)
Date: Wed, 03 Mar 2010 14:49:54 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
Message-ID: <4B8E6902.90106@fx-services.com>

On 2010-03-02 16:02, Jonas Gauffin wrote:

Hi Jonas,

> I got a problem with rejected calls due to 403 (happens sometimes).
> My sip provider found the problem, the nonce used by FS is sometimes
> incorrect (I got the trace from my sip provider).

Your problem is not a bug. The problem is that your SIP provider has two
proxies (that you probably find via SRV). Instead of using one gateway
profile that selects either of them, you have to configure both gateways
as separate elements. That way FS will use the nonce it gets in the 407,
instead of mixing them up between the two proxies (who don't have state).

You either use one gateway for outbound and the other one as failover in
the bridge app, or you use mod_distribute to distribute.

/Robin



From phunk0000 at hotmail.com  Wed Mar  3 06:19:37 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 3 Mar 2010 09:19:37 -0500
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
Message-ID: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>

He group confirm feature seems like it should work, but I can't figure out
where in the default dialplan to put this line so that the audio file is
played for everycall made to a local extension. here us the
group_confirm_playback action, I am wondering where it should go.

 

<action application="bridge" data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

 

Here is the local extension portion of the default dialplan where I think it
should go, but I can't figure out exactly where.  Thanks!

 

<extension name="Local_Extension">

                  <condition field="destination_number"
expression="(^\d{3,6}$)">

                    <!-- <action application="javascript"
data="pbxpalintro.js" /> -->

                   

                    <action application="set" data="dialed_extension=$1"/>

                    <action application="export"
data="dialed_extension=$1"/>

                    

                    <action application="bridge"
data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

                   

                    <!-- bind_meta_app can have these args <key> [a|b|ab]
[a|b|o|s] <app> -->

                    <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>

                    <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

                    <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>

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

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

                    <action application="set" data="call_timeout=30"/>

                    <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->

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

                    <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,
NO_ROUTE_DESTINATION"/> -->

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

                    <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>

                    <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>

                    <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>

                    <!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name
} var sip_secure_media)}"/>-->

                    <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

                    <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

                    <action application="answer"/>

                    <action application="sleep" data="1000"/>

                    <action application="voicemail" data="default
${domain_name} ${dialed_extension}"/>

                  </condition>

                </extension>

 

 

 

 

 

 

 

                   

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, March 02, 2010 11:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

 

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks

Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC 

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From brian at freeswitch.org  Wed Mar  3 06:30:35 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 08:30:35 -0600
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
	<BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
Message-ID: <28735474-6C43-4B1C-A815-CEB9CF39128E@freeswitch.org>

you need user/user at domain not user/user%domain,,,, The faq outlines this syntax used for sofia/profiel/user%domain notation.  Please don't confuse the two.

/b

On Mar 3, 2010, at 8:19 AM, Todd wrote:

> <action application="bridge" data="{group_confirm_file=playback /usr/local/freeswitch/sounds/en/pbxpalintro.wav, group_confirm_key=exec,call_timeout=60} user/${dialed_extension}%${domain_name}"/>

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From phunk0000 at hotmail.com  Wed Mar  3 06:37:33 2010
From: phunk0000 at hotmail.com (Todd)
Date: Wed, 3 Mar 2010 09:37:33 -0500
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
	<BLU110-DS11870819E24598D644773AA73A0@phx.gbl>
Message-ID: <BLU110-DS946BAEF42517F230E5CB4A73A0@phx.gbl>

Or maybe there is an easier way to play an audio file to whoever answers the
phone before the call is bridged..  And is it best practice to put these
changes in the default dial plan or somewhere else?  Still kinda new to
this, Thanks.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd
Sent: Wednesday, March 03, 2010 9:20 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

He group confirm feature seems like it should work, but I can't figure out
where in the default dialplan to put this line so that the audio file is
played for everycall made to a local extension. here us the
group_confirm_playback action, I am wondering where it should go.

 

<action application="bridge" data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

 

Here is the local extension portion of the default dialplan where I think it
should go, but I can't figure out exactly where.  Thanks!

 

<extension name="Local_Extension">

                  <condition field="destination_number"
expression="(^\d{3,6}$)">

                    <!-- <action application="javascript"
data="pbxpalintro.js" /> -->

                   

                    <action application="set" data="dialed_extension=$1"/>

                    <action application="export"
data="dialed_extension=$1"/>

                    

                    <action application="bridge"
data="{group_confirm_file=playback
/usr/local/freeswitch/sounds/en/pbxpalintro.wav,
group_confirm_key=exec,call_timeout=60}
user/${dialed_extension}%${domain_name}"/>

                   

                    <!-- bind_meta_app can have these args <key> [a|b|ab]
[a|b|o|s] <app> -->

                    <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>

                    <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m
-%d-%H-%M-%S)}.wav"/>

                    <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>

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

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

                    <action application="set" data="call_timeout=30"/>

                    <!-- <action application="set"
data="sip_exclude_contact=${network_addr}"/> -->

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

                    <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,
NO_ROUTE_DESTINATION"/> -->

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

                    <action application="db"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb
er}"/>

                    <action application="db"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>

                    <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>

                    <!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name
} var sip_secure_media)}"/>-->

                    <action application="db"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

                    <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>

                    <action application="answer"/>

                    <action application="sleep" data="1000"/>

                    <action application="voicemail" data="default
${domain_name} ${dialed_extension}"/>

                  </condition>

                </extension>

 

 

 

 

 

 

 

                   

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, March 02, 2010 11:04 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Play message upon call anwser

 

 

On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com> wrote:

Hey List..  I am trying to setup my dial plan so whenever a call is answered
a recording is played to the person who answered the phone before the call
is connected..  Any help would be great.   Thanks

Are you talking about the group confirm feature?
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

-MC 

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From brent at overthewire.com.au  Tue Mar  2 22:19:35 2010
From: brent at overthewire.com.au (Brent Paddon)
Date: Wed, 3 Mar 2010 16:19:35 +1000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com> 
	<3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
Message-ID: <ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>

I suspect you would be better off with some virtualization platform more
like OpenVZ ('containers' rather than VM's) - and probably without any
hardware
installed (ie SIP to SIP - leave the gateways to a different part of the
network).

We've run some Asterisk instances in Proxmox (
http://pve.proxmox.com/wiki/Main_Page) and while it's early days, it looks
pretty
promising.

Brent

On Wed, Mar 3, 2010 at 3:27 PM, Brian May <brian at microcomaustralia.com.au>wrote:

> On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> > Is anyone running FreeSWITCH virtualized (xen or vmware) in
> > production?  I saw ec2 and xen mentioned a few places.  If so, are
> > there tips or tricks for handle timer issues which affect virtual
> > machines in general, such as recommended divider or clocksource
> > settings?  I'm seeing funny timing issues while testing with audio
> > recording and playback that speed up or slow down.
>
> I tried tdm400+xen DOM0+asterisk years ago, on a single processor
> machine but gave up. zttest returned good results, but analogue
> quality has intermittently very poor quality and it made telephone
> conversations difficult to understand. Especially a problem when
> hourly cron jobs would start on all VMs at exactly the same time, but
> was a problem at other times too. Plus the fact people who used the
> phone more often then me didn't report the problem, so I thought it
> was fine.
>
> Plus the general fact I didn't really like the idea of running
> Asterisk on DOM0, and running it on DOMU seemed to imply even more
> overheard, as DOM0 would have to process the interrupts and pass them
> to the DOMU.
>
> Unfortunately I don't remember what scheduler I used, maybe sedf?
>
> I suspect a multi-processor system would be a lot better suited for
> the task. Also might be OK if you don't use zaptel hardware... I seem
> to remember these cards generate a huge number of interrupts.
>
> While this was with Asterisk, I suspect my results were not specific
> to Asterisk in anyway.
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
--
Brent Paddon

Director | Over the Wire Pty Ltd brent.paddon at overthewire.com.au |
www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0400 2400 54
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From sos at sokhapkin.dyndns.org  Wed Mar  3 07:17:00 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 3 Mar 2010 10:17:00 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<3c5cf5261003022127o6a16b2der84a6711565367026@mail.gmail.com>
	<ad139bd21003022219o1c5cbb00x2428ccfa6bcd8453@mail.gmail.com>
Message-ID: <201003031017.00850.sos@sokhapkin.dyndns.org>

I second that, do not use VMs. I run tens of asterisks under linux vserver on 
the single box, no issues. I do not expect FS will behave differently under 
vserver.

On Wednesday 03 March 2010, Brent Paddon wrote:
> I suspect you would be better off with some virtualization platform more
> like OpenVZ ('containers' rather than VM's) - and probably without any
> hardware
> installed (ie SIP to SIP - leave the gateways to a different part of the
> network).
>
> We've run some Asterisk instances in Proxmox (
> http://pve.proxmox.com/wiki/Main_Page) and while it's early days, it looks
> pretty
> promising.
>
> Brent
>
> On Wed, Mar 3, 2010 at 3:27 PM, Brian May 
<brian at microcomaustralia.com.au>wrote:
> > On 3 March 2010 16:04, Rob Forman <rob4manhere at gmail.com> wrote:
> > > Is anyone running FreeSWITCH virtualized (xen or vmware) in
> > > production?  I saw ec2 and xen mentioned a few places.  If so, are
> > > there tips or tricks for handle timer issues which affect virtual
> > > machines in general, such as recommended divider or clocksource
> > > settings?  I'm seeing funny timing issues while testing with audio
> > > recording and playback that speed up or slow down.
> >
> > I tried tdm400+xen DOM0+asterisk years ago, on a single processor
> > machine but gave up. zttest returned good results, but analogue
> > quality has intermittently very poor quality and it made telephone
> > conversations difficult to understand. Especially a problem when
> > hourly cron jobs would start on all VMs at exactly the same time, but
> > was a problem at other times too. Plus the fact people who used the
> > phone more often then me didn't report the problem, so I thought it
> > was fine.
> >
> > Plus the general fact I didn't really like the idea of running
> > Asterisk on DOM0, and running it on DOMU seemed to imply even more
> > overheard, as DOM0 would have to process the interrupts and pass them
> > to the DOMU.
> >
> > Unfortunately I don't remember what scheduler I used, maybe sedf?
> >
> > I suspect a multi-processor system would be a lot better suited for
> > the task. Also might be OK if you don't use zaptel hardware... I seem
> > to remember these cards generate a huge number of interrupts.
> >
> > While this was with Asterisk, I suspect my results were not specific
> > to Asterisk in anyway.
> > --
> > Brian May <brian at microcomaustralia.com.au>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org





From javieraristizabal at gmail.com  Wed Mar  3 07:19:26 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Wed, 3 Mar 2010 10:19:26 -0500
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<ec2ce17e1003030042u5b0cca70u6b8e0a4952a64f47@mail.gmail.com>
Message-ID: <b6f11f301003030719gb64ec72k5f38c031f1ce9049@mail.gmail.com>

Hi.. I'm interesting too..

Thanks :)

On Wed, Mar 3, 2010 at 3:42 AM, jay binks <jaybinks at gmail.com> wrote:

> im VERY interested in this...
> ( and the use of NoSQL in general with FS )
>
> please can you email over what you have, or post it to jira.
>
> J
>
> On Wed, Mar 3, 2010 at 4:38 PM, Dome Charoenyost <dome at tel.co.th> wrote:
>
>> Dear All,
>>     I use nibllebill + postgresql for longtime. it' work wel when
>> concurrent about 500 calls..  Now i have 1200 calls .  postgresql  got
>> many problem and slow. so i have 2 choice.
>> 1. upgrade hardware , add more memory may be 32 GB and change HDD to
>> SSD (or IOdrive ).
>> 2. split DB some path to NoSQL (I'm testing memcachedb,redis,tokyo tyrant)
>> I choose 2. and use tokyo tyrant. because it's persistant and support
>> memcache API.
>>  Today i create new module base on nibblebill but use memcacahe API
>> connect to tokyo tyrant. it's look fast. and work without any problem.
>>  If someone want to use my module I'l acttach to JIRA and create WIKI
>> page.
>>
>> Please let's me know if you want.
>>
>> BG
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar  3 08:19:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Mar 2010 08:19:11 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003030819ld02171ub6bc4cbc191eeb94@mail.gmail.com>

Hello all. We have a nice agenda today:

http://bit.ly/9zDiBq

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

Talk to you all soon,
Michael
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From anthony.minessale at gmail.com  Wed Mar  3 08:23:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Mar 2010 10:23:47 -0600
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
In-Reply-To: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
References: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
Message-ID: <191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>

try again 16880 or higher

On Wed, Mar 3, 2010 at 2:06 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> was any timing related change done recently in FS?
> I have updated to trunk r. 16873 (was just some 200 revisions behind) and
> now I'm seeing several messages like this when I try as little as 5 calls
> per second:
> 2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle
> Error! 111
>
> When this happens, FS refuses the related call with "503 Maximum Calls In
> Progress" although, I'm using default config files (sessions-per-second=30).
> So it seems to me FS is wrongly aggregating incoming calls on the same
> interval.
>
> I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
> FS never showed this before in any of them.
>
> regards,
> takeshi
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From chris at fowler.cc  Wed Mar  3 09:17:02 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 3 Mar 2010 12:17:02 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>

Hi Rob,

We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily supporting 100 Polycom users, conference bridges etc.  Been running for over a year.  We also use FlowRoute as our PSTN->SIP Interface, and also Skype Business SIP.  Our users are scattered across the globe - so having the box sitting on AWS' network infrastructure is key to avoiding issues with latency, jitter, packet loss (i.e. I don't think we could afford the connectivity AWS gives us if we had to provision this in-house).

There were no special tricks; you do need to modify/override the following with the box's Elastic IP (EIP).

modify conf/vars.xml and update
  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>

  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=flowroute.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>

conf/sip_profiles/internal.xml
 <param name="aggressive-nat-detection" value="true"/>
 <param name="multiple-registrations" value="true"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="NDLB-received-in-nat-reg-contact" value="true"/>
 <param name="NDLB-force-rport" value="true"/>
 <param name="NDLB-broken-auth-hash" value="true"/>
 <param name="enable-timer" value="false"/>

conf/sip_profiles/external.xml
 <param name="aggressive-nat-detection" value="true"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="NDLB-force-rport" value="true"/>

conf/autoload/switch.conf.xml
    <param name="rtp-start-port" value="16384"/>
    <param name="rtp-end-port" value="32768"/>


Cheers, Chris.
--
RightScale, Inc.



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
Sent: Tuesday, March 02, 2010 9:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Virtualized FreeSWITCH

Hi all,

Is anyone running FreeSWITCH virtualized (xen or vmware) in
production?  I saw ec2 and xen mentioned a few places.  If so, are
there tips or tricks for handle timer issues which affect virtual
machines in general, such as recommended divider or clocksource
settings?  I'm seeing funny timing issues while testing with audio
recording and playback that speed up or slow down.

I have tried both paravirt and hvm.  The default xen kernel has a
timer of 250, so I tried to hvm instead using the stock 5.3 kernel
with the 1khz timer but didn't see much improvement.

All of this was tested on 64-bit CentOS 5.3.
- Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=250
- Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
   CONFIG_HZ=1000


I'm sure physical hosts are the de-facto best practice, but are
virtual machines not worth it at all?

Cheers,
Rob

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



From mayamatakeshi at gmail.com  Wed Mar  3 09:22:17 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 4 Mar 2010 02:22:17 +0900
Subject: [Freeswitch-users] "Throttle Error" after updating to trunk
In-Reply-To: <191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>
References: <15b9404e1003030006o29e0c729w9c78ccfb2d7620b9@mail.gmail.com>
	<191c3a031003030823v5cd17166wc8b14e6f81af17e3@mail.gmail.com>
Message-ID: <15b9404e1003030922p716b1c0cj3999b1d4ff5cf2ad@mail.gmail.com>

It is OK now, Anthony.
Thanks.


On Thu, Mar 4, 2010 at 1:23 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try again 16880 or higher
>
> On Wed, Mar 3, 2010 at 2:06 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> Hello,
>> was any timing related change done recently in FS?
>> I have updated to trunk r. 16873 (was just some 200 revisions behind) and
>> now I'm seeing several messages like this when I try as little as 5 calls
>> per second:
>> 2010-03-03 16:55:16.139880 [CRIT] switch_core_session.c:1496 Throttle
>> Error! 111
>>
>> When this happens, FS refuses the related call with "503 Maximum Calls In
>> Progress" although, I'm using default config files (sessions-per-second=30).
>> So it seems to me FS is wrongly aggregating incoming calls on the same
>> interval.
>>
>> I have tested this in a CentOS5.3 64-bit and with a CentOS5.2 32-bit.
>> FS never showed this before in any of them.
>>
>> regards,
>> takeshi
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar  3 09:23:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Mar 2010 09:23:39 -0800
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <87f2f3b91003030923v24deedc1wd034f1c1337c3536@mail.gmail.com>

Chris,

This is good stuff! Would you be willing to compare this to what's on the
wiki and update the wiki as needed? I believe we have some EC2 information
on the wiki...

Thanks,
MC

On Wed, Mar 3, 2010 at 9:17 AM, Chris Fowler <chris at fowler.cc> wrote:

> Hi Rob,
>
> We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily supporting
> 100 Polycom users, conference bridges etc.  Been running for over a year.
>  We also use FlowRoute as our PSTN->SIP Interface, and also Skype Business
> SIP.  Our users are scattered across the globe - so having the box sitting
> on AWS' network infrastructure is key to avoiding issues with latency,
> jitter, packet loss (i.e. I don't think we could afford the connectivity AWS
> gives us if we had to provision this in-house).
>
> There were no special tricks; you do need to modify/override the following
> with the box's Elastic IP (EIP).
>
> modify conf/vars.xml and update
>  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>
>
>  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=flowroute.com
> "/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>
>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>
>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>
>
> conf/sip_profiles/internal.xml
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="multiple-registrations" value="true"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="NDLB-received-in-nat-reg-contact" value="true"/>
>  <param name="NDLB-force-rport" value="true"/>
>  <param name="NDLB-broken-auth-hash" value="true"/>
>  <param name="enable-timer" value="false"/>
>
> conf/sip_profiles/external.xml
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="NDLB-force-rport" value="true"/>
>
> conf/autoload/switch.conf.xml
>    <param name="rtp-start-port" value="16384"/>
>    <param name="rtp-end-port" value="32768"/>
>
>
> Cheers, Chris.
> --
> RightScale, Inc.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Forman
> Sent: Tuesday, March 02, 2010 9:05 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Virtualized FreeSWITCH
>
> Hi all,
>
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production?  I saw ec2 and xen mentioned a few places.  If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings?  I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.
>
> I have tried both paravirt and hvm.  The default xen kernel has a
> timer of 250, so I tried to hvm instead using the stock 5.3 kernel
> with the 1khz timer but didn't see much improvement.
>
> All of this was tested on 64-bit CentOS 5.3.
> - Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
> SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
> Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
> 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=1000
>
>
> I'm sure physical hosts are the de-facto best practice, but are
> virtual machines not worth it at all?
>
> Cheers,
> Rob
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rob4manhere at gmail.com  Wed Mar  3 09:47:35 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 3 Mar 2010 11:47:35 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
References: <8EA5AF34-49E2-4594-91E1-8D05E66FB460@gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <3F2BADF0-DE12-42A8-9F39-2B89FCCB45DD@gmail.com>

Hi Chris,

Thanks for sharing what you're doing.  That's very helpful.  Which AMI  
did you deploy from?  I'm wondering what your ec2 kernel timer is set  
to.

Thanks!
Rob

On Mar 3, 2010, at 11:17 AM, Chris Fowler wrote:

> Hi Rob,
>
> We run FreeSWITCH on AWS' EC2.  A medium EC2 instance is happily  
> supporting 100 Polycom users, conference bridges etc.  Been running  
> for over a year.  We also use FlowRoute as our PSTN->SIP Interface,  
> and also Skype Business SIP.  Our users are scattered across the  
> globe - so having the box sitting on AWS' network infrastructure is  
> key to avoiding issues with latency, jitter, packet loss (i.e. I  
> don't think we could afford the connectivity AWS gives us if we had  
> to provision this in-house).
>
> There were no special tricks; you do need to modify/override the  
> following with the box's Elastic IP (EIP).
>
> modify conf/vars.xml and update
>  <X-PRE-PROCESS cmd="set" data="domain=<YOUR DOMAIN HERE>"/>
>
>  <X-PRE-PROCESS cmd="set" data="default_provider=sip.flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_username=<USER>"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_password=<PASS>"/>
>  <X-PRE-PROCESS cmd="set"  
> data="default_provider_from_domain=flowroute.com"/>
>  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>
>  <X-PRE-PROCESS cmd="set" data="bind_server_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<AWS EIP>"/>
>  <X-PRE-PROCESS cmd="set" data="external_sip_ip=<AWS EIP>"/>
>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=<YOUR NAME>"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=<YOUR CID>"/>
>
> conf/sip_profiles/internal.xml
> <param name="aggressive-nat-detection" value="true"/>
> <param name="multiple-registrations" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> <param name="NDLB-received-in-nat-reg-contact" value="true"/>
> <param name="NDLB-force-rport" value="true"/>
> <param name="NDLB-broken-auth-hash" value="true"/>
> <param name="enable-timer" value="false"/>
>
> conf/sip_profiles/external.xml
> <param name="aggressive-nat-detection" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> <param name="NDLB-force-rport" value="true"/>
>
> conf/autoload/switch.conf.xml
>    <param name="rtp-start-port" value="16384"/>
>    <param name="rtp-end-port" value="32768"/>
>
>
> Cheers, Chris.
> --
> RightScale, Inc.
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Rob Forman
> Sent: Tuesday, March 02, 2010 9:05 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Virtualized FreeSWITCH
>
> Hi all,
>
> Is anyone running FreeSWITCH virtualized (xen or vmware) in
> production?  I saw ec2 and xen mentioned a few places.  If so, are
> there tips or tricks for handle timer issues which affect virtual
> machines in general, such as recommended divider or clocksource
> settings?  I'm seeing funny timing issues while testing with audio
> recording and playback that speed up or slow down.
>
> I have tried both paravirt and hvm.  The default xen kernel has a
> timer of 250, so I tried to hvm instead using the stock 5.3 kernel
> with the 1khz timer but didn't see much improvement.
>
> All of this was tested on 64-bit CentOS 5.3.
> - Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1
> SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP
> Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=250
> - Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan
> 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
>   CONFIG_HZ=1000
>
>
> I'm sure physical hosts are the de-facto best practice, but are
> virtual machines not worth it at all?
>
> Cheers,
> Rob
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From robert.hadley at teotech.com  Wed Mar  3 09:52:15 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Wed, 3 Mar 2010 09:52:15 -0800
Subject: [Freeswitch-users] Delete custom voicemail greetings
In-Reply-To: <7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com>
	<7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
Message-ID: <8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>

Thanks for replying.  I tried make install samples but it didn't work.

 

My original question should have been "How would an individual user delete a
custom voicemail greeting and reset back to the default greeting?"

 

Thanks,

Robert

 

  _____  

From: Madovsky [mailto:infos at madovsky.org] 
Sent: Tuesday, March 02, 2010 6:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

 

make install samples

----- Original Message ----- 

From: Robert <mailto:robert.hadley at teotech.com>  Hadley 

To: freeswitch-users at lists.freeswitch.org 

Sent: Tuesday, March 02, 2010 4:31 PM

Subject: [Freeswitch-users] Delete custom voicemail greetings

 

Hi FS users,

 

How do I delete custom voicemail greetings and go back to the default
greeting? 

 

Thanks,

Robert


  _____  


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

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From infos at madovsky.org  Wed Mar  3 10:03:26 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 3 Mar 2010 13:03:26 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com><7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
	<8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>
Message-ID: <380273FE029E45BCB5D83B505372F27A@MOBILEE1705>

try to loo at the FS sources folder, I'm sure there will be the right .wav for that
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 12:52 PM
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings


  Thanks for replying.  I tried make install samples but it didn't work.

   

  My original question should have been "How would an individual user delete a custom voicemail greeting and reset back to the default greeting?"

   

  Thanks,

  Robert

   


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

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Tuesday, March 02, 2010 6:47 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

   

  make install samples

    ----- Original Message ----- 

    From: Robert Hadley 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Tuesday, March 02, 2010 4:31 PM

    Subject: [Freeswitch-users] Delete custom voicemail greetings

     

    Hi FS users,

     

    How do I delete custom voicemail greetings and go back to the default greeting? 

     

    Thanks,

    Robert


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

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



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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From freeswitch at cartissolutions.com  Wed Mar  3 10:04:26 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Wed, 03 Mar 2010 12:04:26 -0600
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
Message-ID: <4B8EA4AA.4010602@cartissolutions.com>


On 03/03/2010 12:38 AM, Dome Charoenyost wrote:

<snip snip>

>    Today i create new module base on nibblebill but use memcacahe API
> connect to tokyo tyrant. it's look fast. and work without any problem.
>    If someone want to use my module I'l acttach to JIRA and create WIKI page.
>
> Please let's me know if you want.
>
> BG
> Dome C.
>
>    

Maybe it would be even more useful to provide general functionality that 
can be shared amongst multiple components of freeswitch.  That would 
make it all the more useful.  However, seeing that I'm not a big fan of 
the NoSQL data engines, I don't know if this is possible.  In my 
opinion, most NoSQL is just key=>value pairs, and is basically a 
reinvention of the wheel that RDBMS's had taken care of years ago...   
Not meaning to open up a discussion about the virtues of NoSQL (and 
certainly not a flamewar)...


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com





From helmut.kuper at ewetel.de  Wed Mar  3 10:16:23 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 03 Mar 2010 19:16:23 +0100
Subject: [Freeswitch-users] RFC3326 Reason Header field
Message-ID: <4B8EA777.9080207@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I'm looking for a way to add a reason header field (like 'Reason:
SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
send by bridge application when a call times out or was denied.

I would like to set it on demand in dialplan, e.g. when a queue is
calling a queue member and I don't want to have that call in the phone's
missed calls list.

Is there a way in FS to do this, yet?

best regards
helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
hcWP9lUEFxINGpyLXgfDKLU=
=8qjy
-----END PGP SIGNATURE-----



From Russell.Mosemann at cune.org  Wed Mar  3 10:19:09 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 3 Mar 2010 18:19:09 -0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
Message-ID: <20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>

Chris Fowler <chris at fowler.cc> said:

> There were no special tricks; you do need to modify/override the
> following with the box's Elastic IP (EIP).
 
This would be helpful to have in the wiki along with any other tips for
your virtual environment. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From brian at freeswitch.org  Wed Mar  3 10:21:49 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 12:21:49 -0600
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <4B8EA777.9080207@ewetel.de>
References: <4B8EA777.9080207@ewetel.de>
Message-ID: <0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>

FreeSWITCH already does this for you on cancel when the call is answered elsewhere.

/b

On Mar 3, 2010, at 12:16 PM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hello,
> 
> I'm looking for a way to add a reason header field (like 'Reason:
> SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
> send by bridge application when a call times out or was denied.
> 
> I would like to set it on demand in dialplan, e.g. when a queue is
> calling a queue member and I don't want to have that call in the phone's
> missed calls list.
> 
> Is there a way in FS to do this, yet?
> 
> best regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> 
> iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
> hcWP9lUEFxINGpyLXgfDKLU=
> =8qjy
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rupa at rupa.com  Wed Mar  3 10:25:37 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 3 Mar 2010 12:25:37 -0600
Subject: [Freeswitch-users] Nibblebill for NoSQL
In-Reply-To: <4B8EA4AA.4010602@cartissolutions.com>
References: <8ccbff061003022238j50d28546j34d85b05cce40a9f@mail.gmail.com>
	<4B8EA4AA.4010602@cartissolutions.com>
Message-ID: <b39273531003031025l4222b38cvced0b205b042174a@mail.gmail.com>

On Wed, Mar 3, 2010 at 12:04 PM, Yossi Neiman <
freeswitch at cartissolutions.com> wrote:

>
>
> Maybe it would be even more useful to provide general functionality that
> can be shared amongst multiple components of freeswitch.  That would
> make it all the more useful.  However, seeing that I'm not a big fan of
> the NoSQL data engines, I don't know if this is possible.  In my
> opinion, most NoSQL is just key=>value pairs, and is basically a
> reinvention of the wheel that RDBMS's had taken care of years ago...
> Not meaning to open up a discussion about the virtues of NoSQL (and
> certainly not a flamewar)...
>
>
I created a mod_memcache already and I intend to do a mod_redis at some
point.  I'm not a huge fan of NoSQL but there are definite uses for it.

General functionality:  I could be convinced to come up with a general api
for distributed key/value store with mod_memcache and mod_redis providing
implementation.  The problem is that the NoSQL stuff isn't very consistent
so even though both support key/value their behavior can significantly
differ.  eg: updating a value in memcached doesn't bump the expire time but
does in redis.  Also, redis has support for a much more robust set of value
types and operators.

Anyway, a generic distributed key/value api might look like:

dhash <backend> set key value [expire]
dhash <backend> setnx key value [expire]  # only set if it doesn't already
exist
dhash <backend> get key [...] # support multiple keys
dhash <backend> <increment|decrement> key [step]
dhash <backend> del key

where backend would be memcache or redis or some other implementation.

Notice I didn't even touch things like hashing to support sharding,
failover, etc.


>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
-- 
-Rupa
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From jason at jasonjgw.net  Wed Mar  3 10:42:34 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 03 Mar 2010 12:42:34 -0600
Subject: Tienes una invitación de Jason White a ClubWNC
Message-ID: <E1NmtWk-0004pD-Oi@eclipse.websitewelcome.com>

ClubWNC&nbsp;freeswitch-users: Tienes una invitaci?n de Jason White a ClubWNC 	&nbsp; 	Hola  
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2465                           	Jason White ha invitado a freeswitch-users a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From stevendt at primrosebank.net  Wed Mar  3 11:39:49 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 19:39:49 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payload
	check.
Message-ID: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>

Hi,

can someone please advise what the above error message means ?

"[ERR] switch_rtp.c:2196 Failed DTMF payload check."

It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?

regards
Dave

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From brian at freeswitch.org  Wed Mar  3 11:50:42 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 13:50:42 -0600
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payload
	check.
In-Reply-To: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
Message-ID: <59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>

It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.

/b

On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:

>  
> It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?
>  

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From stevendt at primrosebank.net  Wed Mar  3 12:06:53 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:06:53 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF
	payloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
	<59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>
Message-ID: <E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>

Hi,

thanks Brian.

The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

regards
Dave
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 7:50 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


  It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


  /b


  On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



    It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mrene_lists at avgs.ca  Wed Mar  3 12:09:31 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 15:09:31 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF
	payloadcheck.
In-Reply-To: <E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com>
	<59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org>
	<E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
Message-ID: <3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>

I don't recall any rfc2833 changes in the past 2 weeks. I would like  
to see a packet capture of those bogus rtp packets though.

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




On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:

> Hi,
>
> thanks Brian.
>
> The device is a Thomson ST2030 which has a basic listing on the  
> Interop Wiki page, but does not specifically mention rfc2833  
> (although rtp is marked as "not tested"). I have not seen this  
> message before while running the FS version that I'm currently using  
> 16543 - yes, I know that this is a few weeks old and I will update  
> and/or raise a bug once I've upgraded, but it would be nice to know  
> if anyone else sees problems with this phone,
>
> regards
> Dave
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 7:50 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF  
> payloadcheck.
>
> It usually means your device can't do rfc2833 correctly and needs to  
> have a bug opened so they can fix it.
>
> /b
>
> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>
>>
>> It appears in response to keypad events during voicemail retrieval,  
>> the prompts were actioned correctly, but I was wondering what the  
>> message actually means ?
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From stevendt at primrosebank.net  Wed Mar  3 12:23:02 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:23:02 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed
	DTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com>
	<3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
Message-ID: <36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>

Hi Mathieu,

if you tell me how to generate one, I'll do it now ?

regards
Dave
  ----- Original Message ----- 
  From: Mathieu Rene 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:09 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


  I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








  On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


    Hi,

    thanks Brian.

    The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

    regards
    Dave
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 7:50 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


      It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


      /b


      On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



        It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From stevendt at primrosebank.net  Wed Mar  3 12:39:13 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:39:13 -0000
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
Message-ID: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>

Some more info . . . 

The error only seems to occur when a "0" is entered.

Looking at the Phone config, the RFC2833 options are :-

In Band
Out of Band (RFC2833) - currently selected
SIP Info

RTP Payload Type (96-127) - 96 currently selected
RTP DTMP Level (0-63) - 0 Currently selected

Dave
  ----- Original Message ----- 
  From: Dave Stevenson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:23 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


  Hi Mathieu,

  if you tell me how to generate one, I'll do it now ?

  regards
  Dave
    ----- Original Message ----- 
    From: Mathieu Rene 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 03, 2010 8:09 PM
    Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


    I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though. 


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








    On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


      Hi,

      thanks Brian.

      The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

      regards
      Dave
        ----- Original Message -----
        From: Brian West
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 7:50 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


        It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it. 


        /b


        On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



          It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From chris at fowler.cc  Wed Mar  3 12:48:44 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 3 Mar 2010 15:48:44 -0500
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>

Re: Wiki - I'd love to give back to the community;  Does anyone have an objection if I create a "How to use FreeSWITCH on EC2 with RightScale's management dashboard"?

There are some significant differences to the configuration when using RightScale vs. bare bones AWS; and given this is on my employer's time I need to keep it relevant...

The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is a public AMI (you don't need to be a RightScale customer to use it)

Cheers, Chris.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
Sent: Wednesday, March 03, 2010 10:19 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH

Chris Fowler <chris at fowler.cc> said:

> There were no special tricks; you do need to modify/override the
> following with the box's Elastic IP (EIP).
 
This would be helpful to have in the wiki along with any other tips for
your virtual environment. :-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mrene_lists at avgs.ca  Wed Mar  3 12:51:24 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 15:51:24 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
In-Reply-To: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
	<282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
Message-ID: <D4211946-B75E-4A00-8720-8DAB233E936C@avgs.ca>

You can grab one with wireshark.

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




On 3-Mar-10, at 3:39 PM, Dave Stevenson wrote:

> Some more info . . .
>
> The error only seems to occur when a "0" is entered.
>
> Looking at the Phone config, the RFC2833 options are :-
>
> In Band
> Out of Band (RFC2833) - currently selected
> SIP Info
>
> RTP Payload Type (96-127) - 96 currently selected
> RTP DTMP Level (0-63) - 0 Currently selected
>
> Dave
> ----- Original Message -----
> From: Dave Stevenson
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:23 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196  
> FailedDTMFpayloadcheck.
>
> Hi Mathieu,
>
> if you tell me how to generate one, I'll do it now ?
>
> regards
> Dave
> ----- Original Message -----
> From: Mathieu Rene
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:09 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed  
> DTMFpayloadcheck.
>
> I don't recall any rfc2833 changes in the past 2 weeks. I would like  
> to see a packet capture of those bogus rtp packets though.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:
>
>> Hi,
>>
>> thanks Brian.
>>
>> The device is a Thomson ST2030 which has a basic listing on the  
>> Interop Wiki page, but does not specifically mention rfc2833  
>> (although rtp is marked as "not tested"). I have not seen this  
>> message before while running the FS version that I'm currently  
>> using 16543 - yes, I know that this is a few weeks old and I will  
>> update and/or raise a bug once I've upgraded, but it would be nice  
>> to know if anyone else sees problems with this phone,
>>
>> regards
>> Dave
>> ----- Original Message -----
>> From: Brian West
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Wednesday, March 03, 2010 7:50 PM
>> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF  
>> payloadcheck.
>>
>> It usually means your device can't do rfc2833 correctly and needs  
>> to have a bug opened so they can fix it.
>>
>> /b
>>
>> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>>
>>>
>>> It appears in response to keypad events during voicemail  
>>> retrieval, the prompts were actioned correctly, but I was  
>>> wondering what the message actually means ?
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From stevendt at primrosebank.net  Wed Mar  3 12:56:36 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 3 Mar 2010 20:56:36 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<D4211946-B75E-4A00-8720-8DAB233E936C@avgs.ca>
Message-ID: <885F4FFA1CFC4A328862377FA9253AE1@bp1.ad.bp.com>

Ahh,

OK, then I'll need to go and do some work on it. WireShark is not installed on the machine. I've never used it, although I do have a copy, so I'll look at installing in on the server and go from there,

regards
Dave
  ----- Original Message ----- 
  From: Mathieu Rene 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 8:51 PM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  You can grab one with wireshark.


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








  On 3-Mar-10, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


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








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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



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From gmaruzz at celliax.org  Wed Mar  3 13:00:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 3 Mar 2010 22:00:00 +0100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
Message-ID: <7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>

On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
> Re: Wiki - I'd love to give back to the community; ?Does anyone have an objection if I create a "How to use FreeSWITCH on EC2 with RightScale's management dashboard"?
>
> There are some significant differences to the configuration when using RightScale vs. bare bones AWS; and given this is on my employer's time I need to keep it relevant...

Chris,
I know can be more work, but the best would be to let know the
differences too, and how it would be using bare bone AWS. That's also
good for your employer, if he decide to switch back from RightScale...
:)

-giovanni


>
> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is a public AMI (you don't need to be a RightScale customer to use it)
>
> Cheers, Chris.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org
> Sent: Wednesday, March 03, 2010 10:19 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>
> Chris Fowler <chris at fowler.cc> said:
>
>> There were no special tricks; you do need to modify/override the
>> following with the box's Elastic IP (EIP).
>
> This would be helpful to have in the wiki along with any other tips for
> your virtual environment. :-)
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From rob4manhere at gmail.com  Wed Mar  3 13:09:49 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Wed, 3 Mar 2010 15:09:49 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
Message-ID: <25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>

 >  Cheers, Chris.
 >  --
 >  RightScale, Inc.

I doubt they'll be switching any time soon.  I think his employer IS  
RightScale :)


On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:

> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>> Re: Wiki - I'd love to give back to the community;  Does anyone  
>> have an objection if I create a "How to use FreeSWITCH on EC2 with  
>> RightScale's management dashboard"?
>>
>> There are some significant differences to the configuration when  
>> using RightScale vs. bare bones AWS; and given this is on my  
>> employer's time I need to keep it relevant...
>
> Chris,
> I know can be more work, but the best would be to let know the
> differences too, and how it would be using bare bone AWS. That's also
> good for your employer, if he decide to switch back from RightScale...
> :)
>
> -giovanni
>
>
>>
>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is  
>> a public AMI (you don't need to be a RightScale customer to use it)
>>
>> Cheers, Chris.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
>> ] On Behalf Of Russell.Mosemann at cune.org
>> Sent: Wednesday, March 03, 2010 10:19 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>
>> Chris Fowler <chris at fowler.cc> said:
>>
>>> There were no special tricks; you do need to modify/override the
>>> following with the box's Elastic IP (EIP).
>>
>> This would be helpful to have in the wiki along with any other tips  
>> for
>> your virtual environment. :-)
>>
>> --
>> Russell Mosemann
>>
>>
>>
>> ________________________________________________________
>> Concordia University, Nebraska
>> See http://www.cune.edu/ for the latest news and events!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at celliax.org  Wed Mar  3 13:15:38 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 3 Mar 2010 22:15:38 +0100
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com> 
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
Message-ID: <7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>

On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com> wrote:
> ?> ?Cheers, Chris.
> ?> ?--
> ?> ?RightScale, Inc.
>
> I doubt they'll be switching any time soon. ?I think his employer IS
> RightScale :)

Oooops, I'll learn to pay attention to signatures :)


>
>
> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>
>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>> Re: Wiki - I'd love to give back to the community; ?Does anyone
>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>> RightScale's management dashboard"?
>>>
>>> There are some significant differences to the configuration when
>>> using RightScale vs. bare bones AWS; and given this is on my
>>> employer's time I need to keep it relevant...
>>
>> Chris,
>> I know can be more work, but the best would be to let know the
>> differences too, and how it would be using bare bone AWS. That's also
>> good for your employer, if he decide to switch back from RightScale...
>> :)
>>
>> -giovanni
>>
>>
>>>
>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>> a public AMI (you don't need to be a RightScale customer to use it)
>>>
>>> Cheers, Chris.
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>> ] On Behalf Of Russell.Mosemann at cune.org
>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>
>>> Chris Fowler <chris at fowler.cc> said:
>>>
>>>> There were no special tricks; you do need to modify/override the
>>>> following with the box's Elastic IP (EIP).
>>>
>>> This would be helpful to have in the wiki along with any other tips
>>> for
>>> your virtual environment. :-)
>>>
>>> --
>>> Russell Mosemann
>>>
>>>
>>>
>>> ________________________________________________________
>>> Concordia University, Nebraska
>>> See http://www.cune.edu/ for the latest news and events!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From helmut.kuper at ewetel.de  Wed Mar  3 16:20:02 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 04 Mar 2010 01:20:02 +0100
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
References: <4B8EA777.9080207@ewetel.de>
	<0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
Message-ID: <4B8EFCB2.9070709@ewetel.de>

Hi Brian,

does this work for serial calls as well ?


Am 03.03.2010 19:21, schrieb Brian West:
> FreeSWITCH already does this for you on cancel when the call is answered elsewhere.
>
> /b
>
> On Mar 3, 2010, at 12:16 PM, Helmut Kuper wrote:
>
>    
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hello,
>>
>> I'm looking for a way to add a reason header field (like 'Reason:
>> SIP;cause=200;text="Call completed elsewhere"') to SIP CANCEL message
>> send by bridge application when a call times out or was denied.
>>
>> I would like to set it on demand in dialplan, e.g. when a queue is
>> calling a queue member and I don't want to have that call in the phone's
>> missed calls list.
>>
>> Is there a way in FS to do this, yet?
>>
>> best regards
>> helmut
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.7 (MingW32)
>>
>> iD8DBQFLjqd34tZeNddg3dwRAvaTAJ46LCweLzvE1pupvcsX3HVR/XW0MACdE5yr
>> hcWP9lUEFxINGpyLXgfDKLU=
>> =8qjy
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>    




From mgg at giagnocavo.net  Wed Mar  3 16:31:54 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 3 Mar 2010 19:31:54 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670353534F47@mse17be1.mse17.exchange.ms>

Without knowing more about your account and provider, perhaps they have had issues with users getting their accounts hacked, so they added on IP authentication in addition to diget? Or maybe they've had issues with IP-auth people still having DoS type attacks by sending fake INVITEs and using up credit a bit at a time, so they added on digest? Or maybe they're paranoid?

-Michael


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jonas Gauffin
Sent: Wednesday, March 03, 2010 1:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] OT: SIP and IP spoofing

Hello,

My sip gateway provider are using both IP address locking (only my servers IP addresses can use my gateway account) and Digest authentication on every call.
I asked why and they said that the account would be vulnerable to IP spoofing otherwise. Is that possible? I mean, if someone fakes my servers IP address in the packets, shouldn't the responses be sent back to my server and not the one creating the fake packets? Are there any other reasons to use both ip locking and digest authentication?

Regards,
  Jonas
-------------- next part --------------
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From brian at freeswitch.org  Wed Mar  3 16:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 18:32:28 -0600
Subject: [Freeswitch-users] RFC3326 Reason Header field
In-Reply-To: <4B8EFCB2.9070709@ewetel.de>
References: <4B8EA777.9080207@ewetel.de>
	<0986F893-F06D-4223-AD1E-85EA5C9A2102@freeswitch.org>
	<4B8EFCB2.9070709@ewetel.de>
Message-ID: <1FF8C930-82A8-4A62-9E4C-503F1FA7AD62@freeswitch.org>

Yes if the call was to say 10 phones at once... and the person answers they hangup with LOSE_RACE which results in the reason header doing that.

in mod_sofia

            if (cause > 0 && cause < 128) {                                                                                                                     
                switch_snprintf(reason, sizeof(reason), "Q.850;cause=%d;text=\"%s\"", cause, switch_channel_cause2str(cause));                                  
            } else if (cause == SWITCH_CAUSE_PICKED_OFF || cause == SWITCH_CAUSE_LOSE_RACE) {
                                                                                   switch_snprintf(reason, sizeof(reason), "SIP;cause=200;text=\"Call completed elsewhere\"");                                                     
            } else {                                                                                                                                            
                switch_snprintf(reason, sizeof(reason), "%s;cause=%d;text=\"%s\"", tech_pvt->profile->username, cause, switch_channel_cause2str(cause));        
            }                                                                                                                                         


/b


On Mar 3, 2010, at 6:20 PM, Helmut Kuper wrote:

> Hi Brian,
> 
> does this work for serial calls as well ?
> 
> 
> Am 03.03.2010 19:21, schrieb Brian West:
>> FreeSWITCH already does this for you on cancel when the call is answered elsewhere.
>> 
>> /b




From mrene_lists at avgs.ca  Wed Mar  3 16:34:51 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 3 Mar 2010 19:34:51 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
Message-ID: <692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>

Theorically you can spoof the network's ip address as long as you set  
the contact header to where the reply should be sent.

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




On 3-Mar-10, at 3:07 AM, Jonas Gauffin wrote:

> Hello,
>
> My sip gateway provider are using both IP address locking (only my  
> servers IP addresses can use my gateway account) and Digest  
> authentication on every call.
> I asked why and they said that the account would be vulnerable to IP  
> spoofing otherwise. Is that possible? I mean, if someone fakes my  
> servers IP address in the packets, shouldn't the responses be sent  
> back to my server and not the one creating the fake packets? Are  
> there any other reasons to use both ip locking and digest  
> authentication?
>
> Regards,
>   Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dujinfang at gmail.com  Wed Mar  3 17:04:08 2010
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 4 Mar 2010 09:04:08 +0800
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
Message-ID: <23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>

we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
pretty well. Of course no much load.

2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com> wrote:
>> ?> ?Cheers, Chris.
>> ?> ?--
>> ?> ?RightScale, Inc.
>>
>> I doubt they'll be switching any time soon. ?I think his employer IS
>> RightScale :)
>
> Oooops, I'll learn to pay attention to signatures :)
>
>
>>
>>
>> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>
>>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>>> Re: Wiki - I'd love to give back to the community; ?Does anyone
>>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>>> RightScale's management dashboard"?
>>>>
>>>> There are some significant differences to the configuration when
>>>> using RightScale vs. bare bones AWS; and given this is on my
>>>> employer's time I need to keep it relevant...
>>>
>>> Chris,
>>> I know can be more work, but the best would be to let know the
>>> differences too, and how it would be using bare bone AWS. That's also
>>> good for your employer, if he decide to switch back from RightScale...
>>> :)
>>>
>>> -giovanni
>>>
>>>
>>>>
>>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>>> a public AMI (you don't need to be a RightScale customer to use it)
>>>>
>>>> Cheers, Chris.
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>> ] On Behalf Of Russell.Mosemann at cune.org
>>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>>
>>>> Chris Fowler <chris at fowler.cc> said:
>>>>
>>>>> There were no special tricks; you do need to modify/override the
>>>>> following with the box's Elastic IP (EIP).
>>>>
>>>> This would be helpful to have in the wiki along with any other tips
>>>> for
>>>> your virtual environment. :-)
>>>>
>>>> --
>>>> Russell Mosemann
>>>>
>>>>
>>>>
>>>> ________________________________________________________
>>>> Concordia University, Nebraska
>>>> See http://www.cune.edu/ for the latest news and events!
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mgg at giagnocavo.net  Wed Mar  3 17:25:31 2010
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 3 Mar 2010 20:25:31 -0500
Subject: [Freeswitch-users] OT: SIP and IP spoofing
In-Reply-To: <692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>
References: <d021d3be1003030007u47cd7e6fn39999246f0bd410c@mail.gmail.com>
	<692F0F79-A066-4CC8-833B-B202117CDBB8@avgs.ca>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C670353534F74@mse17be1.mse17.exchange.ms>

Correct me if I'm wrong; it's fortunately been a while since I've had to deal with SIP details. This attack would depend on the strength of the To tag, right? Hopefully they would append ;received to Vias, so the responses would go to the spoofed IP and not the attacker's IP in the Via. The attacker would have to be able to calculate the To tag to construct a valid ACK. 

But that does depend on received being added and secure To tags - hopefully a security-conscious provider would check their side for that.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Wednesday, March 03, 2010 5:35 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] OT: SIP and IP spoofing

Theorically you can spoof the network's ip address as long as you set the contact header to where the reply should be sent.

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




On 3-Mar-10, at 3:07 AM, Jonas Gauffin wrote:

> Hello,
>
> My sip gateway provider are using both IP address locking (only my 
> servers IP addresses can use my gateway account) and Digest 
> authentication on every call.
> I asked why and they said that the account would be vulnerable to IP 
> spoofing otherwise. Is that possible? I mean, if someone fakes my 
> servers IP address in the packets, shouldn't the responses be sent 
> back to my server and not the one creating the fake packets? Are there 
> any other reasons to use both ip locking and digest authentication?
>
> Regards,
>   Jonas
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org


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



From delorenzodesign at gmail.com  Wed Mar  3 19:04:19 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 3 Mar 2010 22:04:19 -0500
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback doesn't seem to get 	called
In-Reply-To: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
Message-ID: <5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>

Has anyone experienced issues with vmd or dtmf with Verizon wireless?  I
have another script that doesn't even register digits pressed -- it's like
no keys are pressed at all.

I've checked the FS log and it doesn't seem to log anything with regard to
the mod_vmd other than an indication of MIN_TIME of 8000 (not sure what that
does exactly).

Does anyone have any sample scripts available other than what's on the FS
wiki?

On Mon, Mar 1, 2010 at 10:17 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I've got the following Lua script working in a sense, but the InputCallback
> never seems to get called while the file is being streamed to the call
> recipient.  I've tried moving the "vmd start" command and set input callback
> around a bit, but to no avail.  I'm testing this against a cell phone voice
> mailbox (Verizon).
>
>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>>  number_to_call = argv[1]
>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>
>> voicemail_detected = false;
>>
>> function onInput(s, type, obj)
>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>> *************\n");
>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>> session:getVariable("vmd_detect") .. " *************\n");
>>
>>      if(type == "event" and voicemail_detected == false) then
>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>> MACHINE DETECTED *************\n");
>>         voicemail_detected = true;
>>         return "break";
>>     end
>> end
>>
>> function playbackMessage()
>>     sleep_time = 1000;
>>     if(voicemail_detected) then
>>        sleep_time = 2500;
>>     end
>>     -- sleep a second
>>     session:sleep(sleep_time);
>>     -- play a file
>>     session:streamFile(message_to_play);
>> end
>>
>> function notify()
>>     session =
>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>> .. number_to_call)
>>
>>
>>
>>     if(session:ready()) then
>>         -- answer the call
>>         session:answer();
>>         session:setInputCallback("onInput", "");
>>         session:execute("vmd","start");
>>
>>         playbackMessage();
>>         if(voicemail_detected) then
>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>             playbackMessage();
>>         end
>>
>>         freeswitch.consoleLog("notice", "********* hanging up session
>> **********\n");
>>         -- hangup
>>         session:hangup();
>>     end
>> end
>>
>> notify();
>>
>>
>> freeswitch.consoleLog("info","########################################################\n\n");
>>
>
> --
> Michael De Lorenzo
>
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From christian.bourke1 at gmail.com  Wed Mar  3 19:37:08 2010
From: christian.bourke1 at gmail.com (Bilbo)
Date: Wed, 03 Mar 2010 21:37:08 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1s4-0001px-NS@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2896                           	Bilbo ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From dujinfang at gmail.com  Wed Mar  3 19:37:15 2010
From: dujinfang at gmail.com (seven)
Date: Wed, 03 Mar 2010 21:37:15 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895                           	seven ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From msc at freeswitch.org  Wed Mar  3 19:39:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 03 Mar 2010 21:39:43 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1uZ-0002WX-Tu@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2887                           	Michael Collins ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From kristian.kielhofner at gmail.com  Wed Mar  3 19:41:15 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 03 Mar 2010 21:41:15 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2882                           	Kristian Kielhofner ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From bjbrashier at gmail.com  Wed Mar  3 19:42:05 2010
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 03 Mar 2010 21:42:05 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn1wr-0003Er-7J@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2871                           	Bradley Brashier ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From brian at freeswitch.org  Wed Mar  3 20:30:16 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 22:30:16 -0600
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Tienes_una_invitaci=F3n__a_Club?=
	=?iso-8859-1?q?WNC?=
In-Reply-To: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>
References: <E1Nn1w3-00030N-NR@eclipse.websitewelcome.com>
Message-ID: <1CDF95CD-5331-41E8-8B7B-9073AB6D0F7A@freeswitch.org>

WTF is going on here.

Did someone hit google or something?  REALLY?

This clubwnc people need to be beat with a sledge hammer.

/b




From brian at freeswitch.org  Wed Mar  3 20:47:40 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Mar 2010 22:47:40 -0600
Subject: [Freeswitch-users] TEST
Message-ID: <6C78AD58-771B-41FC-8F93-0D2634C44682@freeswitch.org>

TEST

/b




From christian.bourke1 at gmail.com  Wed Mar  3 20:37:44 2010
From: christian.bourke1 at gmail.com (Bilbo)
Date: Wed, 03 Mar 2010 22:37:44 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn2oi-0000vz-8g@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2896                           	Bilbo ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From infos at madovsky.org  Wed Mar  3 22:09:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:09:30 -0500
Subject: [Freeswitch-users] svn 16892 compilation on Fedora10 64bits
Message-ID: <2A95130D21884F8B988302547CA5AB45@MOBILEE1705>

Hello,

just to say that svn version 16892 compiles
well now and CPU load is ok (xeon)
excellent job !

Regards

F
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From infos at madovsky.org  Wed Mar  3 22:31:35 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:31:35 -0500
Subject: [Freeswitch-users] registration status
Message-ID: <A7363AFFB2024D2A92060998A1A91036@MOBILEE1705>

Is the registration status in DB ?

thanks

Franck
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From infos at madovsky.org  Wed Mar  3 22:33:52 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 01:33:52 -0500
Subject: [Freeswitch-users] registration status
Message-ID: <E9A73C7FD69D41A58DBEEC6E621F8DC9@MOBILEE1705>

Forget my question I found the answer 

http://wiki.freeswitch.org/wiki/ODBC#ODBC_in_the_core

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 1:31 AM
  Subject: registration status


  Is the registration status in DB ?

  thanks

  Franck
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From dujinfang at gmail.com  Wed Mar  3 22:46:23 2010
From: dujinfang at gmail.com (seven)
Date: Thu, 04 Mar 2010 00:46:23 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4pD-0003Bv-CU@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895                           	seven ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From msc at freeswitch.org  Wed Mar  3 22:47:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 04 Mar 2010 00:47:12 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4q0-0003LZ-K4@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2887                           	Michael Collins ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From kristian.kielhofner at gmail.com  Wed Mar  3 22:47:40 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 04 Mar 2010 00:47:40 -0600
Subject: Tienes una invitación  a ClubWNC
Message-ID: <E1Nn4qS-0003Rz-Nb@eclipse.websitewelcome.com>

ClubWNC&nbsp;Tienes una invitaci?n  a ClubWNC 	&nbsp; 	Hola  !!!
He creado un perfil en ClubWNC donde puedo hacer listas de regalos, publicar rese?as sobre productos, ver a mis amigos de Facebook  y lo que les gusta, lo que han comprado, y lo que quieren que les regalen, y quiero agregarte a mis amigos para que puedas verlo. Para ello, necesitas registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.              	        Para registrarte en ClubWNC, sigue este enlace:http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2882                           	Kristian Kielhofner ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para borrar tu nombre de la lista de personas suscritas.




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From jonas.gauffin at gmail.com  Wed Mar  3 23:01:54 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 4 Mar 2010 08:01:54 +0100
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <4B8E6902.90106@fx-services.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<4B8E6902.90106@fx-services.com>
Message-ID: <d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>

You are correct, I'm using DNSSRV.

imho it's a bug when FS is not using the same nonce in the response as it
received in the request. It might be a design flaw only that only appears
when running against multiple servers through DNSSRV, but it's still a bug.

I'm sure it's easy to fix for Anthony with his m4d sk1llz ;)

On Wed, Mar 3, 2010 at 2:49 PM, Robin Vleij <viper at fx-services.com> wrote:

> On 2010-03-02 16:02, Jonas Gauffin wrote:
>
> Hi Jonas,
>
> > I got a problem with rejected calls due to 403 (happens sometimes).
> > My sip provider found the problem, the nonce used by FS is sometimes
> > incorrect (I got the trace from my sip provider).
>
> Your problem is not a bug. The problem is that your SIP provider has two
> proxies (that you probably find via SRV). Instead of using one gateway
> profile that selects either of them, you have to configure both gateways
> as separate elements. That way FS will use the nonce it gets in the 407,
> instead of mixing them up between the two proxies (who don't have state).
>
> You either use one gateway for outbound and the other one as failover in
> the bridge app, or you use mod_distribute to distribute.
>
> /Robin
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Thu Mar  4 00:20:10 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 03:20:10 -0500
Subject: [Freeswitch-users] Delete custom voicemail greetings
References: <4D58851467464E15801979AC635DAA00@greyhawk.tonecommander.com><7315F3D9A8C54944B2D8699811DA85D4@MOBILEE1705>
	<8F957DA2497F4E238E254B455B6DA49D@greyhawk.tonecommander.com>
Message-ID: <4DEA455178544A028AFCA7D21644E42E@MOBILEE1705>

sorry make samples
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 03, 2010 12:52 PM
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings


  Thanks for replying.  I tried make install samples but it didn't work.

   

  My original question should have been "How would an individual user delete a custom voicemail greeting and reset back to the default greeting?"

   

  Thanks,

  Robert

   


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

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Tuesday, March 02, 2010 6:47 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Delete custom voicemail greetings

   

  make install samples

    ----- Original Message ----- 

    From: Robert Hadley 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Tuesday, March 02, 2010 4:31 PM

    Subject: [Freeswitch-users] Delete custom voicemail greetings

     

    Hi FS users,

     

    How do I delete custom voicemail greetings and go back to the default greeting? 

     

    Thanks,

    Robert


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From jaybinks at gmail.com  Thu Mar  4 00:38:27 2010
From: jaybinks at gmail.com (jay binks)
Date: Thu, 4 Mar 2010 18:38:27 +1000
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a difference )
Message-ID: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>

has anyone done any work on getting SNMP monitoring of freeswitch ??

from looking at some other emails about this, it seems that there is not a
heap of desire ( from developers point of view )
to have a FS SNMP module.   Ive also noticed most requests for this type of
monitoring are directed to ESL.

so my thought,  has anyone done any work on building a script / module for
NET-SNMP ( http://net-snmp.sourceforge.net/ )
that uses ESL to fetch its data.

so we would run snmpd on our FS boxes, with the FS-ESL module ( or script )
loaded into snmpd.
from there SNMP requests would be handled by snmpd and the data would then
be pulled over ESL directly from freeswitch.

this seems like a decent solution that would not risk stability of FS by
introducing yet another module.

so I guess my question here is, has anyone done something like this ?
what do you all think of this idea ?
is there anyone who would like to contribute to this idea or offer other
suggestions.

oh yea, and why SNMP..  I duno...
SNMP is just so easy to integrate into other monitoring systems.


-- 
Sincerely

Jay
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From testeador01 at gmail.com  Thu Mar  4 05:57:42 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 4 Mar 2010 08:57:42 -0500
Subject: [Freeswitch-users] TEST
Message-ID: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>

you failed!!  O_O
after the messages from clubWNC, I'm not getting any more list messages on
my e-mail inbox, is it normal?
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From brian at freeswitch.org  Thu Mar  4 06:40:15 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 08:40:15 -0600
Subject: [Freeswitch-users] Testing
Message-ID: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>

Testing

/b




From abrownworth at bandwidth.com  Wed Mar  3 13:53:49 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Wed, 3 Mar 2010 16:53:49 -0500
Subject: [Freeswitch-users] Splitting a Call
Message-ID: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>

Hello fellow FreeSWITCH users,

I'm using FreeSWITCH Version 1.0.trunk (16883) and I'm using the event
socket to break an in-progress call into two independently controlled legs.
I want to be able to work with each leg completely independently - hanging
up one without affecting the other. Per the docs:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_transfer I'm using
uuid_transfer with an inline lua script to work with one leg without hanging
up on the other. I'm setting "hangup_after_bridge" to false on both channels
and invoking uuid_transfer with the -bleg option but as soon as I transfer,
FreeSWITCH hangs up on one of the legs.

Here's what I'm sending on the event socket. (everything gets an "OK"
response from FreeSWITCH)

Set hangup_after_bridge:


api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a hangup_after_bridge
false
api uuid_setvar 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 hangup_after_bridge
false

And then do the uuid_transfer:

api uuid_transfer 53d37581-1f90-44bf-860a-addbc8430e3a -bleg
'lua:callee.lua' inline
api uuid_transfer 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 -bleg
'lua:caller.lua' inline

The callee leg (53d37581-1f90-44bf-860a-addbc8430e3a) is immediately hung up
and the caller leg runs fine. Here's the log messages: (6505551212 is the
caller and 5165551212 is the callee)

2010-03-03 21:29:56.086319 [DEBUG] switch_ivr.c:1441 (sofia/external/+
16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.086319 [NOTICE] switch_ivr.c:1447 Transfer
sofia/external/+16505551212 at 192.168.27.72 to inline[lua:callee.lua at public]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/external/+16505551212 at 192.168.27.72]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_rtp.c:1626 Send middle packet for
[1] ts=91040 dur=1760/1760/2440 seq=7493
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
DONE [sofia/external/+15165551212]
2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.092461 [NOTICE] switch_ivr_bridge.c:637 Hangup
sofia/external/+15165551212 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2010-03-03 21:29:56.092461 [DEBUG] switch_channel.c:2071 Send signal
sofia/external/+15165551212 [KILL]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:351
(sofia/external/+15165551212) State EXCHANGE_MEDIA going to sleep
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+15165551212) Running State Change CS_HANGUP
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
(sofia/external/+15165551212) State HANGUP
2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:411 Channel
sofia/external/+15165551212 hanging up, cause: NORMAL_CLEARING
2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:454 Sending BYE to
sofia/external/+15165551212
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:46
sofia/external/+15165551212 Standard HANGUP, cause: NORMAL_CLEARING
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
(sofia/external/+15165551212) State HANGUP going to sleep
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:333
(sofia/external/+15165551212) State Change CS_HANGUP -> CS_REPORTING
2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+15165551212) Running State Change CS_REPORTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:590
(sofia/external/+15165551212) State REPORTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING
2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:140 sofia/external/+
16505551212 at 192.168.27.72 SOFIA ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:77
sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:119
(sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
CS_EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:226 sofia/external/+
16505551212 at 192.168.27.72 SOFIA EXECUTE
2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:157
sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(callee.lua)
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:53
sofia/external/+15165551212 Standard REPORTING, cause: NORMAL_CLEARING
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:590
(sofia/external/+15165551212) State REPORTING going to sleep
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:327
(sofia/external/+15165551212) State Change CS_REPORTING -> CS_DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+15165551212 [BREAK]
2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1161 Session 24
(sofia/external/+15165551212) Locked, Waiting on external entities
2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1179 Session 24
(sofia/external/+15165551212) Ended
2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/external/+15165551212 [CS_DESTROY]
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:428
(sofia/external/+15165551212) Running State Change CS_DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
(sofia/external/+15165551212) State DESTROY
2010-03-03 21:29:56.095544 [DEBUG] mod_sofia.c:338
sofia/external/+15165551212 SOFIA DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:60
sofia/external/+15165551212 Standard DESTROY
2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
(sofia/external/+15165551212) State DESTROY going to sleep
2010-03-03 21:29:56.127189 [DEBUG] switch_ivr.c:1441 (sofia/external/+
16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:638 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.127189 [NOTICE] switch_ivr.c:1447 Transfer
sofia/external/+16505551212 at 192.168.27.72 to inline[lua:caller.lua]
2010-03-03 21:29:56.875077 [DEBUG] switch_core_media_bug.c:360 Attaching BUG
to sofia/external/+16505551212 at 192.168.27.72
2010-03-03 21:29:56.909780 [DEBUG] switch_core_media_bug.c:442 Removing BUG
from sofia/external/+16505551212 at 192.168.27.72
2010-03-03 21:29:56.910802 [ERR] inline:1  Session is not active!
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING
2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:140 sofia/external/+
16505551212 at 192.168.27.72 SOFIA ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:77
sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:119
(sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
CS_EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/+16505551212 at 192.168.27.72 [BREAK]
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
(sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
(sofia/external/+16505551212 at 192.168.27.72) Running State Change CS_EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
(sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:226 sofia/external/+
16505551212 at 192.168.27.72 SOFIA EXECUTE
2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:157
sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(caller.lua)
2010-03-03 21:29:56.911830 [NOTICE] switch_cpp.cpp:1142 ***********
Caller.lua - Success: 53d37581-1f90-44bf-860a-addbc8430e3a ***********
2010-03-03 21:29:56.911830 [ERR] switch_core_session.c:357 Could not locate
channel type 53d37581-1f90-44bf-860a-addbc8430e3a
2010-03-03 21:29:56.911830 [ERR] switch_ivr_originate.c:2389 Cannot create
outgoing channel of type [53d37581-1f90-44bf-860a-addbc8430e3a] cause:
[CHAN_NOT_IMPLEMENTED]
2010-03-03 21:29:56.911830 [DEBUG] switch_ivr_originate.c:3187 Originate
Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2010-03-03 21:29:56.911830 [ERR] switch_cpp.cpp:618 session is not
initalized
2010-03-03 21:29:57.032095 [DEBUG] switch_ivr_play_say.c:1156 Codec
Activated L16 at 8000hz 1 channels 20ms

How can I get the callee leg to stay up? Is this not what -bleg is supposed
to do? Am I doing something wrong here? I've tried parking both legs and
invoking the lua on the channels but one of the channels still drops and the
uuid is no-longer available in the event socket.

Is there some other preferred way to split a call apart?

Thanks.

-Anders
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From abrownworth at bandwidth.com  Wed Mar  3 14:56:37 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Wed, 3 Mar 2010 17:56:37 -0500
Subject: [Freeswitch-users] Splitting a Call
In-Reply-To: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>
References: <dbad22fd1003031353g25945cc1nc1cd3172003b767f@mail.gmail.com>
Message-ID: <dbad22fd1003031456k631a4237tea2663cc03ceb776@mail.gmail.com>

Answering my own question, parking the call with "uuid_transfer <uuid> -both
'park' inline" first keeps the other leg busy and things won't hang up.

-Anders

On Wed, Mar 3, 2010 at 4:53 PM, Anders Brownworth <abrownworth at bandwidth.com
> wrote:

> Hello fellow FreeSWITCH users,
>
> I'm using FreeSWITCH Version 1.0.trunk (16883) and I'm using the event
> socket to break an in-progress call into two independently controlled legs.
> I want to be able to work with each leg completely independently - hanging
> up one without affecting the other. Per the docs:
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_transfer I'm using
> uuid_transfer with an inline lua script to work with one leg without hanging
> up on the other. I'm setting "hangup_after_bridge" to false on both channels
> and invoking uuid_transfer with the -bleg option but as soon as I transfer,
> FreeSWITCH hangs up on one of the legs.
>
> Here's what I'm sending on the event socket. (everything gets an "OK"
> response from FreeSWITCH)
>
> Set hangup_after_bridge:
>
>
> api uuid_setvar 53d37581-1f90-44bf-860a-addbc8430e3a hangup_after_bridge
> false
> api uuid_setvar 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 hangup_after_bridge
> false
>
> And then do the uuid_transfer:
>
> api uuid_transfer 53d37581-1f90-44bf-860a-addbc8430e3a -bleg
> 'lua:callee.lua' inline
> api uuid_transfer 9d1a4f2d-2ce6-40e5-ac84-62ad8c2d4654 -bleg
> 'lua:caller.lua' inline
>
> The callee leg (53d37581-1f90-44bf-860a-addbc8430e3a) is immediately hung
> up and the caller leg runs fine. Here's the log messages: (6505551212 is the
> caller and 5165551212 is the callee)
>
> 2010-03-03 21:29:56.086319 [DEBUG] switch_ivr.c:1441 (sofia/external/+
> 16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
> 2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.086319 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.086319 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/external/+16505551212 at 192.168.27.72 to inline[lua:callee.lua at public]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/external/+16505551212 at 192.168.27.72]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_rtp.c:1626 Send middle packet for
> [1] ts=91040 dur=1760/1760/2440 seq=7493
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:565 BRIDGE THREAD
> DONE [sofia/external/+15165551212]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_ivr_bridge.c:585 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.092461 [NOTICE] switch_ivr_bridge.c:637 Hangup
> sofia/external/+15165551212 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_channel.c:2071 Send signal
> sofia/external/+15165551212 [KILL]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:351
> (sofia/external/+15165551212) State EXCHANGE_MEDIA going to sleep
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+15165551212) Running State Change CS_HANGUP
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/+15165551212) State HANGUP
> 2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:411 Channel
> sofia/external/+15165551212 hanging up, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.092461 [DEBUG] mod_sofia.c:454 Sending BYE to
> sofia/external/+15165551212
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:46
> sofia/external/+15165551212 Standard HANGUP, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/+15165551212) State HANGUP going to sleep
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/+15165551212) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.092461 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+15165551212) Running State Change CS_REPORTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/+15165551212) State REPORTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:140 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:77
> sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:119
> (sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] mod_sofia.c:226 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA EXECUTE
> 2010-03-03 21:29:56.093481 [DEBUG] switch_core_state_machine.c:157
> sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
> EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(callee.lua)
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:53
> sofia/external/+15165551212 Standard REPORTING, cause: NORMAL_CLEARING
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/+15165551212) State REPORTING going to sleep
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/+15165551212) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+15165551212 [BREAK]
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_session.c:1161 Session 24
> (sofia/external/+15165551212) Locked, Waiting on external entities
> 2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1179 Session 24
> (sofia/external/+15165551212) Ended
> 2010-03-03 21:29:56.095544 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/+15165551212 [CS_DESTROY]
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/+15165551212) Running State Change CS_DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/+15165551212) State DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] mod_sofia.c:338
> sofia/external/+15165551212 SOFIA DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:60
> sofia/external/+15165551212 Standard DESTROY
> 2010-03-03 21:29:56.095544 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/+15165551212) State DESTROY going to sleep
> 2010-03-03 21:29:56.127189 [DEBUG] switch_ivr.c:1441 (sofia/external/+
> 16505551212 at 192.168.27.72) State Change CS_EXECUTE -> CS_ROUTING
> 2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.127189 [DEBUG] switch_core_session.c:638 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.127189 [NOTICE] switch_ivr.c:1447 Transfer
> sofia/external/+16505551212 at 192.168.27.72 to inline[lua:caller.lua]
> 2010-03-03 21:29:56.875077 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/external/+16505551212 at 192.168.27.72
> 2010-03-03 21:29:56.909780 [DEBUG] switch_core_media_bug.c:442 Removing BUG
> from sofia/external/+16505551212 at 192.168.27.72
> 2010-03-03 21:29:56.910802 [ERR] inline:1  Session is not active!
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE going to sleep
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:140 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:77
> sofia/external/+16505551212 at 192.168.27.72 Standard ROUTING
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:119
> (sofia/external/+16505551212 at 192.168.27.72) State Change CS_ROUTING ->
> CS_EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/+16505551212 at 192.168.27.72 [BREAK]
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/+16505551212 at 192.168.27.72) State ROUTING going to sleep
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/+16505551212 at 192.168.27.72) Running State Change
> CS_EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:348
> (sofia/external/+16505551212 at 192.168.27.72) State EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] mod_sofia.c:226 sofia/external/+
> 16505551212 at 192.168.27.72 SOFIA EXECUTE
> 2010-03-03 21:29:56.910802 [DEBUG] switch_core_state_machine.c:157
> sofia/external/+16505551212 at 192.168.27.72 Standard EXECUTE
> EXECUTE sofia/external/+16505551212 at 192.168.27.72 lua(caller.lua)
> 2010-03-03 21:29:56.911830 [NOTICE] switch_cpp.cpp:1142 ***********
> Caller.lua - Success: 53d37581-1f90-44bf-860a-addbc8430e3a ***********
> 2010-03-03 21:29:56.911830 [ERR] switch_core_session.c:357 Could not locate
> channel type 53d37581-1f90-44bf-860a-addbc8430e3a
> 2010-03-03 21:29:56.911830 [ERR] switch_ivr_originate.c:2389 Cannot create
> outgoing channel of type [53d37581-1f90-44bf-860a-addbc8430e3a] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2010-03-03 21:29:56.911830 [DEBUG] switch_ivr_originate.c:3187 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2010-03-03 21:29:56.911830 [ERR] switch_cpp.cpp:618 session is not
> initalized
> 2010-03-03 21:29:57.032095 [DEBUG] switch_ivr_play_say.c:1156 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> How can I get the callee leg to stay up? Is this not what -bleg is supposed
> to do? Am I doing something wrong here? I've tried parking both legs and
> invoking the lua on the channels but one of the channels still drops and the
> uuid is no-longer available in the event socket.
>
> Is there some other preferred way to split a call apart?
>
> Thanks.
>
> -Anders
>
>
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From brian at freeswitch.org  Thu Mar  4 06:52:01 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 08:52:01 -0600
Subject: [Freeswitch-users] SUCCESS!!!
Message-ID: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>

I have to say PYTHON SUCKS... ie Mailman is evil.

/b




From testa at voicetechnology.com.br  Thu Mar  4 06:55:24 2010
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Thu, 4 Mar 2010 11:55:24 -0300
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
Message-ID: <9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>

I used to do tests on conferencing using linode.com, which is Xen VMs
without issues, although there were not much load on it.

On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:

> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
> pretty well. Of course no much load.
>
> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
> wrote:
> >>  >  Cheers, Chris.
> >>  >  --
> >>  >  RightScale, Inc.
> >>
> >> I doubt they'll be switching any time soon.  I think his employer IS
> >> RightScale :)
> >
> > Oooops, I'll learn to pay attention to signatures :)
> >
> >
> >>
> >>
> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
> >>
> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
> >>>> RightScale's management dashboard"?
> >>>>
> >>>> There are some significant differences to the configuration when
> >>>> using RightScale vs. bare bones AWS; and given this is on my
> >>>> employer's time I need to keep it relevant...
> >>>
> >>> Chris,
> >>> I know can be more work, but the best would be to let know the
> >>> differences too, and how it would be using bare bone AWS. That's also
> >>> good for your employer, if he decide to switch back from RightScale...
> >>> :)
> >>>
> >>> -giovanni
> >>>
> >>>
> >>>>
> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
> >>>> a public AMI (you don't need to be a RightScale customer to use it)
> >>>>
> >>>> Cheers, Chris.
> >>>>
> >>>> -----Original Message-----
> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org
> >>>> ] On Behalf Of Russell.Mosemann at cune.org
> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
> >>>> To: freeswitch-users at lists.freeswitch.org
> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
> >>>>
> >>>> Chris Fowler <chris at fowler.cc> said:
> >>>>
> >>>>> There were no special tricks; you do need to modify/override the
> >>>>> following with the box's Elastic IP (EIP).
> >>>>
> >>>> This would be helpful to have in the wiki along with any other tips
> >>>> for
> >>>> your virtual environment. :-)
> >>>>
> >>>> --
> >>>> Russell Mosemann
> >>>>
> >>>>
> >>>>
> >>>> ________________________________________________________
> >>>> Concordia University, Nebraska
> >>>> See http://www.cune.edu/ for the latest news and events!
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>> Sincerely,
> >>>
> >>> Giovanni Maruzzelli
> >>> Cell : +39-347-2665618
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mustafa.pk at gmail.com  Thu Mar  4 07:08:34 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 4 Mar 2010 20:08:34 +0500
Subject: [Freeswitch-users] Testing
In-Reply-To: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>
References: <6B4B6A58-5A68-4EC6-B13D-A5DAA9C66519@freeswitch.org>
Message-ID: <8213d6071003040708i790de267g8863980c2d7e0f5f@mail.gmail.com>

echo 123

On Thu, Mar 4, 2010 at 7:40 PM, Brian West <brian at freeswitch.org> wrote:
> Testing
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com



From paul at apcl.us  Thu Mar  4 07:11:33 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 04 Mar 2010 10:11:33 -0500
Subject: [Freeswitch-users] how to see registration info for a particular
	user?
Message-ID: <4B8FCDA5.6000700@apcl.us>

I am familiar with the console command:
     sofia status profile internal reg
to see the registration info for all currently registered users.

Is there a modification to this command that will limit the result to 
just one specific user registered user?

   Thanks,
   Paul




From gmaruzz at celliax.org  Thu Mar  4 07:22:01 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 4 Mar 2010 16:22:01 +0100
Subject: [Freeswitch-users] Your Future Order with 75% off retail
Message-ID: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>

Testing

:P

-giovanni
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From brian at freeswitch.org  Thu Mar  4 07:27:56 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 09:27:56 -0600
Subject: [Freeswitch-users] Your Future Order with 75% off retail
In-Reply-To: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>
References: <7b197bef1003040722j66c92e92sfc9ea03bc5c32546@mail.gmail.com>
Message-ID: <EB62BDFE-D71B-4A3A-BF09-CB285EEE9BB7@freeswitch.org>

You crazy!

/b
PS: mirror image of me.

On Mar 4, 2010, at 9:22 AM, Giovanni Maruzzelli wrote:

> Testing
> 
> :P
> 
> -giovanni
> -- 
> Sincerely,




From infos at madovsky.org  Thu Mar  4 08:49:00 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 11:49:00 -0500
Subject: [Freeswitch-users] SUCCESS!!!
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
Message-ID: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>

Mailman, I ll get your scalp 

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 04, 2010 9:52 AM
Subject: [Freeswitch-users] SUCCESS!!!


>I have to say PYTHON SUCKS... ie Mailman is evil.
> 
> /b
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Mar  4 09:13:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 11:13:59 -0600
Subject: [Freeswitch-users] http://jira.freeswitch.org/browse/ESL-33
Message-ID: <78042E0E-4E1B-41A9-885F-867DD621FCE2@freeswitch.org>

Any java web heads out there wanna take a smack at this one?

/b




From anthony.minessale at gmail.com  Thu Mar  4 09:17:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 11:17:38 -0600
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
Message-ID: <191c3a031003040917p47f87a73hc559a5d8633b31fc@mail.gmail.com>

Let's upgrade to Angry Dog


On Thu, Mar 4, 2010 at 10:49 AM, Madovsky <infos at madovsky.org> wrote:

> Mailman, I ll get your scalp
>
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, March 04, 2010 9:52 AM
> Subject: [Freeswitch-users] SUCCESS!!!
>
>
> >I have to say PYTHON SUCKS... ie Mailman is evil.
> >
> > /b
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From m.sobkow at marketelsystems.com  Thu Mar  4 09:22:07 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 11:22:07 -0600
Subject: [Freeswitch-users] How to play back a .wav file from Erlang?
Message-ID: <4B8FEC3F.5060102@marketelsystems.com>

I have the UUID of a parked call.  I've waited n seconds for an operator 
to answer that call, but none are available so I want to play a message 
to the customer.  How do I do that from Erlang?  I've tried the 
following, but it doesn't seem to actually play the file, and I'm not 
getting any useful information in fs_cli about the playback attempt that 
would indicate why it's not working.

bgapi( eval, "uuid:1f6d7a61-fd1a-4efd-bf84-31c4eea652c2 playback 
/opt/freeswitch/sounds/en/us/callie/misc/8000/sorry.wav" )

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




From m.sobkow at marketelsystems.com  Thu Mar  4 10:36:23 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 12:36:23 -0600
Subject: [Freeswitch-users] Call keeps hanging up
Message-ID: <4B8FFDA7.50708@marketelsystems.com>

I'm having some difficulty with Freeswitch/Erlang processing.  An 
operator dials in and registers by entering a PIN code, which triggers 
an Erlang event handler callback and puts the call into park.  This works.

A customer calls in, which triggers another Erlang event handler 
callback and puts the call into park.  That works, too.

Another thread of Erlang checks the operator and customer queues, and 
uses uuid_bridge to join the two calls together.  That also works.

What _doesn't_ work is that when the customer hangs up, I want the 
operator A leg to go back into a park state, and it's hanging up 
instead.  One suspicion I have is that because there is no dialplan 
associated with the call (it's controlled by Erlang), there is no 
dialplan to "continue" by setting hangup_after_bridge=false.  Either 
that or maybe I need to set the variables on the new UUID returned by 
the uuid_bridge command (though I thought that UUID was just the UUID 
assigned to the command, not a new call identifier.)

BTW, "pbx" is just an OTP rewrite of pieces of the standard 
"freeswitch.erl" code.


    pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
++ " hangup_after_bridge false" ),
    pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
++ " park_after_bridge true" ),

    BridgeResult = pbx:api( uuid_bridge, 
Operator#pbx_operator_registry.operator_uuid ++ " " ++ 
Customer#pbx_cust_queue.uuid ),

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




From andrew at hijacked.us  Thu Mar  4 10:47:55 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Mar 2010 13:47:55 -0500
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B8FFDA7.50708@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
Message-ID: <20100304184755.GK1751@hijacked.us>

On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
> I'm having some difficulty with Freeswitch/Erlang processing.  An 
> operator dials in and registers by entering a PIN code, which triggers 
> an Erlang event handler callback and puts the call into park.  This works.
> 
> A customer calls in, which triggers another Erlang event handler 
> callback and puts the call into park.  That works, too.
> 
> Another thread of Erlang checks the operator and customer queues, and 
> uses uuid_bridge to join the two calls together.  That also works.
> 
> What _doesn't_ work is that when the customer hangs up, I want the 
> operator A leg to go back into a park state, and it's hanging up 
> instead.  One suspicion I have is that because there is no dialplan 
> associated with the call (it's controlled by Erlang), there is no 
> dialplan to "continue" by setting hangup_after_bridge=false.  Either 
> that or maybe I need to set the variables on the new UUID returned by 
> the uuid_bridge command (though I thought that UUID was just the UUID 
> assigned to the command, not a new call identifier.)
> 
> BTW, "pbx" is just an OTP rewrite of pieces of the standard 
> "freeswitch.erl" code.
> 
> 
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
> ++ " hangup_after_bridge false" ),
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid 
> ++ " park_after_bridge true" ),
>

Try using park_after_bridge and catch the park event or
transfer_after_bridge and set the dialplan location to transfer the call
to.

Andrew



From anthony.minessale at gmail.com  Thu Mar  4 11:12:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 13:12:50 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>

"Hello this is the United States Calling are we reaching... *click* See!, he
keeps hanging up, is there supposed to be someone there besides your wife
sir?"


On Thu, Mar 4, 2010 at 12:47 PM, Andrew Thompson <andrew at hijacked.us> wrote:

> On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
> > I'm having some difficulty with Freeswitch/Erlang processing.  An
> > operator dials in and registers by entering a PIN code, which triggers
> > an Erlang event handler callback and puts the call into park.  This
> works.
> >
> > A customer calls in, which triggers another Erlang event handler
> > callback and puts the call into park.  That works, too.
> >
> > Another thread of Erlang checks the operator and customer queues, and
> > uses uuid_bridge to join the two calls together.  That also works.
> >
> > What _doesn't_ work is that when the customer hangs up, I want the
> > operator A leg to go back into a park state, and it's hanging up
> > instead.  One suspicion I have is that because there is no dialplan
> > associated with the call (it's controlled by Erlang), there is no
> > dialplan to "continue" by setting hangup_after_bridge=false.  Either
> > that or maybe I need to set the variables on the new UUID returned by
> > the uuid_bridge command (though I thought that UUID was just the UUID
> > assigned to the command, not a new call identifier.)
> >
> > BTW, "pbx" is just an OTP rewrite of pieces of the standard
> > "freeswitch.erl" code.
> >
> >
> >     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> > ++ " hangup_after_bridge false" ),
> >     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> > ++ " park_after_bridge true" ),
> >
>
> Try using park_after_bridge and catch the park event or
> transfer_after_bridge and set the dialplan location to transfer the call
> to.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jerry.richards at teotech.com  Thu Mar  4 12:19:40 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 12:19:40 -0800
Subject: [Freeswitch-users] Dialing **<extension>
Message-ID: <613B61547E7C43FFB4978276BDD146D9@greyhawk.tonecommander.com>

I used to be able to dial "**<extension>" to pickup a ringing call (v.
1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
this no longer works the same.

Is this feature still available under some other star-code?

Best Regards,
Jerry




From tculjaga at gmail.com  Thu Mar  4 12:44:23 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 4 Mar 2010 21:44:23 +0100
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
Message-ID: <65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>

On Thu, Mar 4, 2010 at 9:38 AM, jay binks <jaybinks at gmail.com> wrote:

> has anyone done any work on getting SNMP monitoring of freeswitch ??
>
> from looking at some other emails about this, it seems that there is not a
> heap of desire ( from developers point of view )
> to have a FS SNMP module.   Ive also noticed most requests for this type of
> monitoring are directed to ESL.
>

because there is no need :)


>
> so my thought,  has anyone done any work on building a script / module for
> NET-SNMP ( http://net-snmp.sourceforge.net/ )
> that uses ESL to fetch its data.
>
>
what i'm doing is just make a script to call fs_cli -x command

e.g.

tculjaga at nemesis:~$ /usr/local/freeswitch/bin/fs_cli -x "status"
UP 0 years, 0 days, 0 hours, 0 minutes, 8 seconds, 470 milliseconds, 62
microseconds
0 session(s) since startup
0 session(s) 0/30
1000 session(s) max


and bind it to an OID...


I'm even firing SIPP with to check end-to-end FS sanity.



> so we would run snmpd on our FS boxes, with the FS-ESL module ( or script )
> loaded into snmpd.
> from there SNMP requests would be handled by snmpd and the data would then
> be pulled over ESL directly from freeswitch.
>
> this seems like a decent solution that would not risk stability of FS by
> introducing yet another module.
>
> so I guess my question here is, has anyone done something like this ?
>

yes...





> what do you all think of this idea ?
> is there anyone who would like to contribute to this idea or offer other
> suggestions.
>
>
it is just a script ... quite easy to setup...

prepare your script:

[tculjaga@ bin]$ cat activeChannels.sh
#!/bin/sh

CHANNELS=`/usr/local/freeswitch/bin/fs_cli -x "show channels count"
|/bin/grep total | /bin/awk {'print $1'}`
/bin/echo $CHANNELS
exit $CHANNELS
[tculjaga at l01sipindir2 bin]$
[tculjaga at l01sipindir2 bin]$
[tculjaga at l01sipindir2 bin]$ cat currentCallsPerSec.sh
#!/bin/sh



in snmpd.conf add this:

exec activeChannels /usr/local/bin/activeChannels.sh
exec checkCallFlowCPS /usr/local/bin/sipgen/sipgen.sh 1234 1 10




> oh yea, and why SNMP..  I duno...
> SNMP is just so easy to integrate into other monitoring systems.
>
>
because every NMS uses it .. maybe thats why :)


>
>

> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Thu Mar  4 12:48:06 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 14:48:06 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <4B901C86.7040806@marketelsystems.com>

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From jimthomasembedded at yahoo.com  Thu Mar  4 13:10:55 2010
From: jimthomasembedded at yahoo.com (Jim Thomas)
Date: Thu, 4 Mar 2010 13:10:55 -0800 (PST)
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <407604F34A7540C3B971AABC4D952060@MOBILEE1705>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
Message-ID: <43146.92621.qm@web44811.mail.sp1.yahoo.com>

> PYTHON SUCKS

That's like saying:

? Asterisk sucks.
? Asterisk is written in C language.
? Therefore, C language sucks.

:-)

?
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, March 04, 2010 9:52 AM
Subject: [Freeswitch-users] SUCCESS!!!


>I have to say PYTHON SUCKS... ie Mailman is evil.
> 
> /b



      
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From brian at freeswitch.org  Thu Mar  4 13:22:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 15:22:17 -0600
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <43146.92621.qm@web44811.mail.sp1.yahoo.com>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
	<43146.92621.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <B9F8B4A9-368B-42D5-B30C-E49756710264@freeswitch.org>

You can create shitty art with paint... doesn't mean the paint sucks.

/b

On Mar 4, 2010, at 3:10 PM, Jim Thomas wrote:

> > PYTHON SUCKS
>  
> That's like saying:
>  
>   Asterisk sucks.
>   Asterisk is written in C language.
>   Therefore, C language sucks.
>  
> :-)
> 

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From christian.loeschenkohl at xpirio.com  Thu Mar  4 13:43:31 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 04 Mar 2010 22:43:31 +0100
Subject: [Freeswitch-users] how to see registration info for a
 particular user?
In-Reply-To: <4B8FCDA5.6000700@apcl.us>
References: <4B8FCDA5.6000700@apcl.us>
Message-ID: <4B902983.9080207@xpirio.com>

hi

simply add the user id to the end of the command
like "sofia status profile internal reg 123456"
where 123456 is the id of the user you want to see

br

Paul Levin wrote:

> I am familiar with the console command:
>      sofia status profile internal reg
> to see the registration info for all currently registered users.
> 
> Is there a modification to this command that will limit the result to 
> just one specific user registered user?
> 
>    Thanks,
>    Paul
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From jerry.richards at teotech.com  Thu Mar  4 14:14:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 14:14:30 -0800
Subject: [Freeswitch-users] Dialing **<extension>
Message-ID: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>


By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry




-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Thursday, March 04, 2010 12:20 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Dialing **<extension>

I used to be able to dial "**<extension>" to pickup a ringing call (v.
1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
this no longer works the same.

Is this feature still available under some other star-code?

Best Regards,
Jerry




From jaybinks at gmail.com  Thu Mar  4 14:25:57 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 08:25:57 +1000
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com> 
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
Message-ID: <ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>

HUH, well there you go..   exactly what I was after...
someone has done exactly what I was thinking.

I know you've sent it to me by email, but lets get this in the wiki.
Ill put this up from your email, but if you have more id encourage you to
share what youve done.

Jay


On Fri, Mar 5, 2010 at 6:44 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
>
> On Thu, Mar 4, 2010 at 9:38 AM, jay binks <jaybinks at gmail.com> wrote:
>
>> has anyone done any work on getting SNMP monitoring of freeswitch ??
>>
>> from looking at some other emails about this, it seems that there is not a
>> heap of desire ( from developers point of view )
>> to have a FS SNMP module.   Ive also noticed most requests for this type
>> of monitoring are directed to ESL.
>>
>
> because there is no need :)
>
>
>>
>> so my thought,  has anyone done any work on building a script / module for
>> NET-SNMP ( http://net-snmp.sourceforge.net/ )
>> that uses ESL to fetch its data.
>>
>>
> what i'm doing is just make a script to call fs_cli -x command
>
> e.g.
>
> tculjaga at nemesis:~$ /usr/local/freeswitch/bin/fs_cli -x "status"
> UP 0 years, 0 days, 0 hours, 0 minutes, 8 seconds, 470 milliseconds, 62
> microseconds
> 0 session(s) since startup
> 0 session(s) 0/30
> 1000 session(s) max
>
>
> and bind it to an OID...
>
>
> I'm even firing SIPP with to check end-to-end FS sanity.
>
>
>
>> so we would run snmpd on our FS boxes, with the FS-ESL module ( or script
>> ) loaded into snmpd.
>> from there SNMP requests would be handled by snmpd and the data would then
>> be pulled over ESL directly from freeswitch.
>>
>> this seems like a decent solution that would not risk stability of FS by
>> introducing yet another module.
>>
>> so I guess my question here is, has anyone done something like this ?
>>
>
> yes...
>
>
>
>
>
>> what do you all think of this idea ?
>> is there anyone who would like to contribute to this idea or offer other
>> suggestions.
>>
>>
> it is just a script ... quite easy to setup...
>
> prepare your script:
>
> [tculjaga@ bin]$ cat activeChannels.sh
> #!/bin/sh
>
> CHANNELS=`/usr/local/freeswitch/bin/fs_cli -x "show channels count"
> |/bin/grep total | /bin/awk {'print $1'}`
> /bin/echo $CHANNELS
> exit $CHANNELS
> [tculjaga at l01sipindir2 bin]$
> [tculjaga at l01sipindir2 bin]$
> [tculjaga at l01sipindir2 bin]$ cat currentCallsPerSec.sh
> #!/bin/sh
>
>
>
> in snmpd.conf add this:
>
> exec activeChannels /usr/local/bin/activeChannels.sh
> exec checkCallFlowCPS /usr/local/bin/sipgen/sipgen.sh 1234 1 10
>
>
>
>
>> oh yea, and why SNMP..  I duno...
>> SNMP is just so easy to integrate into other monitoring systems.
>>
>>
> because every NMS uses it .. maybe thats why :)
>
>
>>
>>
>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From m.sobkow at marketelsystems.com  Thu Mar  4 14:35:32 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 04 Mar 2010 16:35:32 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304184755.GK1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
Message-ID: <4B9035B4.6030901@marketelsystems.com>

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From msc at freeswitch.org  Thu Mar  4 14:58:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 14:58:36 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>

On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, it appears that the "**<extension>" ringing call pickup feature
> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
> of
> the following scenario:
>
> 1) 1059 calls 5381
> 2) 5381 rings
> 3) 5402 dials "**5381"
>
> In the bad case (http://pastebin.freeswitch.org/12325), all calls
> disconnect
> following above scenario.  In the good case
> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>
> Best Regards,
> Jerry
>

I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC
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From gorand at donevtechconsulting.com  Thu Mar  4 14:59:37 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 16:59:37 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0ca201cabbee$5ee077a0$1ca166e0$@com>

Howdy Freeswitch community; 

I was wondering if the developers are any closer to the release of 1.05
version. Please don't point me to an e-mail link :) Just inquiring as to
whether there is a firm close date of 1.05. 

Thanks. 
Goran




From msc at freeswitch.org  Thu Mar  4 15:00:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:00:12 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B9035B4.6030901@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
Message-ID: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>

On Thu, Mar 4, 2010 at 2:35 PM, Mark Sobkow <m.sobkow at marketelsystems.com>wrote:

>  Andrew Thompson wrote:
>
> On Thu, Mar 04, 2010 at 12:36:23PM -0600, Mark Sobkow wrote:
>
>
>  I'm having some difficulty with Freeswitch/Erlang processing.  An
> operator dials in and registers by entering a PIN code, which triggers
> an Erlang event handler callback and puts the call into park.  This works.
>
> A customer calls in, which triggers another Erlang event handler
> callback and puts the call into park.  That works, too.
>
> Another thread of Erlang checks the operator and customer queues, and
> uses uuid_bridge to join the two calls together.  That also works.
>
> What _doesn't_ work is that when the customer hangs up, I want the
> operator A leg to go back into a park state, and it's hanging up
> instead.  One suspicion I have is that because there is no dialplan
> associated with the call (it's controlled by Erlang), there is no
> dialplan to "continue" by setting hangup_after_bridge=false.  Either
> that or maybe I need to set the variables on the new UUID returned by
> the uuid_bridge command (though I thought that UUID was just the UUID
> assigned to the command, not a new call identifier.)
>
> BTW, "pbx" is just an OTP rewrite of pieces of the standard
> "freeswitch.erl" code.
>
>
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> ++ " hangup_after_bridge false" ),
>     pbx:api( uuid_setvar, Operator#pbx_operator_registry.operator_uuid
> ++ " park_after_bridge true" ),
>
>
>
>  Try using park_after_bridge and catch the park event or
> transfer_after_bridge and set the dialplan location to transfer the call
> to.
>
> Andrew
>
>
>
>  What a relief.  "transfer_after_bridge" worked!  Plus it provided an
> obvious way for me to collect the call result code, fire it back to Erlang,
> and leave the operator waiting for the next incoming call. :)
>
> "Another satisfied customer!" :D Now it's time to throw some of that
Canadian colored money at Tony's paypal account. ;)
BTW, what kind of solution are you developing?
-MC
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From freeswitch at cartissolutions.com  Thu Mar  4 15:04:26 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:04:26 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>	<20100304184755.GK1751@hijacked.us>
	<191c3a031003041112q7762f50bya40087db8a967397@mail.gmail.com>
Message-ID: <4B903C7A.7030902@cartissolutions.com>


On 03/04/2010 01:12 PM, Anthony Minessale wrote:
> "Hello this is the United States Calling are we reaching... *click* 
> See!, he keeps hanging up, is there supposed to be someone there 
> besides your wife sir?"
>
>

"And it's a man answering..."   *ring ring...    ring ring...*



From anthony.minessale at gmail.com  Thu Mar  4 15:06:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:06:19 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
Message-ID: <191c3a031003041506r4bad1d75pd28839b3c334dea1@mail.gmail.com>

1) You are supposed to report bugs to jira http://jira.freeswitch.org
2) You are not using svn trunk so the line numbers don't match for
debugging.
3) #2 is a prerequisite for #1


On Thu, Mar 4, 2010 at 4:14 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>
> By the way, it appears that the "**<extension>" ringing call pickup feature
> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
> of
> the following scenario:
>
> 1) 1059 calls 5381
> 2) 5381 rings
> 3) 5402 dials "**5381"
>
> In the bad case (http://pastebin.freeswitch.org/12325), all calls
> disconnect
> following above scenario.  In the good case
> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>
> Best Regards,
> Jerry
>
>
>
>
> -----Original Message-----
> From: Jerry Richards [mailto:jerry.richards at teotech.com]
> Sent: Thursday, March 04, 2010 12:20 PM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: Dialing **<extension>
>
> I used to be able to dial "**<extension>" to pickup a ringing call (v.
> 1.0.5pre9) from a different extension.  Since I upgraded to 1.0.5-20100223,
> this no longer works the same.
>
> Is this feature still available under some other star-code?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar  4 15:07:34 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:07:34 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <4B903D36.2040407@cartissolutions.com>


On 03/04/2010 04:59 PM, Goran Donev wrote:
> Howdy Freeswitch community;
>
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05.
>
> Thanks.
> Goran
>    

"We are delighted that your curiosity of our planet continues unabated.  
And we're like to assure you that the two nuclear-armed missiles are 
merely a courtesy detail..."

Standard response:  It's in the oven baking right now.  The muffin man 
will take it out as soon as it's ready...


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com





From brian at freeswitch.org  Thu Mar  4 15:07:58 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:07:58 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <C278735C-3EA9-46B0-835F-1AF0E241DC34@freeswitch.org>

Its getting there.  Please see jira... and keep an eye out the best thing anyone can do at this point is TEST TEST TEST TEST.  ;)

/b

On Mar 4, 2010, at 4:59 PM, Goran Donev wrote:

> Howdy Freeswitch community; 
> 
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05. 
> 
> Thanks. 
> Goran
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Thu Mar  4 15:09:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:09:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <0ca201cabbee$5ee077a0$1ca166e0$@com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
Message-ID: <191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>

No we have no deadline, we have spent countless hours preparing and we still
have a small list of packaging concerns.  Feel free to download the code
from http://latest.freeswitch.org and rest assured it *is* 1.0.5

On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> Howdy Freeswitch community;
>
> I was wondering if the developers are any closer to the release of 1.05
> version. Please don't point me to an e-mail link :) Just inquiring as to
> whether there is a firm close date of 1.05.
>
> Thanks.
> Goran
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar  4 15:15:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 17:15:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
Message-ID: <191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>

You know, that's the only question you have asked on this list.

I went back and searched for your name.

And you have asked it like 12 times and have been answered the same each
time.

What is the obsession with an arbitrary tag?


On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> No we have no deadline, we have spent countless hours preparing and we
> still have a small list of packaging concerns.  Feel free to download the
> code from http://latest.freeswitch.org and rest assured it *is* 1.0.5
>
> On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <
> gorand at donevtechconsulting.com> wrote:
>
>> Howdy Freeswitch community;
>>
>> I was wondering if the developers are any closer to the release of 1.05
>> version. Please don't point me to an e-mail link :) Just inquiring as to
>> whether there is a firm close date of 1.05.
>>
>> Thanks.
>> Goran
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar  4 15:30:23 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 04 Mar 2010 17:30:23 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>	<0ca201cabbee$5ee077a0$1ca166e0$@com>	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
Message-ID: <4B90428F.1000408@cartissolutions.com>


On 03/04/2010 05:15 PM, Anthony Minessale wrote:
> You know, that's the only question you have asked on this list.
>
> I went back and searched for your name.
>
> And you have asked it like 12 times and have been answered the same 
> each time.
>
> What is the obsession with an arbitrary tag?

Maybe we should name the 1.0.5 official release as the "Donev Release"...


>
> On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     No we have no deadline, we have spent countless hours preparing
>     and we still have a small list of packaging concerns.  Feel free
>     to download the code from http://latest.freeswitch.org and rest
>     assured it *is* 1.0.5
>
>     On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev
>     <gorand at donevtechconsulting.com
>     <mailto:gorand at donevtechconsulting.com>> wrote:
>
>         Howdy Freeswitch community;
>
>         I was wondering if the developers are any closer to the
>         release of 1.05
>         version. Please don't point me to an e-mail link :) Just
>         inquiring as to
>         whether there is a firm close date of 1.05.
>
>         Thanks.
>         Goran
>
>


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com


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From andrew at hijacked.us  Thu Mar  4 15:40:50 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 4 Mar 2010 18:40:50 -0500
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
Message-ID: <20100304234050.GL1751@hijacked.us>

On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:

> > "Another satisfied customer!" :D Now it's time to throw some of that
> Canadian colored money at Tony's paypal account. ;)

Hey! I wrote the erlang module and provided the solution to the problem
;)

Andrew



From gorand at donevtechconsulting.com  Thu Mar  4 15:41:17 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 17:41:17 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 47
In-Reply-To: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0cd601cabbf4$30f000d0$92d00270$@com>

It has been baking for a while. What's the timer on the oven say, how much
more till it's done.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
freeswitch-users-request at lists.freeswitch.org
Sent: Thursday, March 04, 2010 5:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 45, Issue 47

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of FreeSWITCH-users digest..."




From brian at freeswitch.org  Thu Mar  4 15:41:21 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:41:21 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B90428F.1000408@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>	<0ca201cabbee$5ee077a0$1ca166e0$@com>	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
Message-ID: <12464591-4662-494A-AC23-ED322F8F6B94@freeswitch.org>

HAHAHA NO! :P

/b

On Mar 4, 2010, at 5:30 PM, Yossi Neiman wrote:

> Maybe we should name the 1.0.5 official release as the "Donev Release"...




From jaybinks at gmail.com  Thu Mar  4 15:42:25 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 09:42:25 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B903D36.2040407@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
Message-ID: <ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>

who is the muffin man !??

yuuummm muffins... I Want some...
fresh yummy muffins...

is BKW the muffin man ?

J


> Standard response:  It's in the oven baking right now.  The muffin man
> will take it out as soon as it's ready...
>
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From msc at freeswitch.org  Thu Mar  4 15:53:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:53:45 -0800
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 47
In-Reply-To: <0cd601cabbf4$30f000d0$92d00270$@com>
References: <mailman.22.1267744057.19150.freeswitch-users@lists.freeswitch.org>
	<0cd601cabbf4$30f000d0$92d00270$@com>
Message-ID: <87f2f3b91003041553o1309f27dtf81573e6613775f2@mail.gmail.com>

On Thu, Mar 4, 2010 at 3:41 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> It has been baking for a while. What's the timer on the oven say, how much
> more till it's done.
>
> The answer is.... "It depends." For many of us, current SVN is already
baked and ready for action. I've been using SVN on several machines for the
past month and all has been well. For me, SVN has been "1.0.5" for at least
four weeks. The stuff that they're working on is really oddball bugs from
goofy use cases and dealing with bugs from phone manufacturers...

-MC
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From msc at freeswitch.org  Thu Mar  4 15:54:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Mar 2010 15:54:23 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304234050.GL1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<20100304234050.GL1751@hijacked.us>
Message-ID: <87f2f3b91003041554y1c7fbe51h956b525ab6dcee67@mail.gmail.com>

On Thu, Mar 4, 2010 at 3:40 PM, Andrew Thompson <andrew at hijacked.us> wrote:

> On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:
>
> > > "Another satisfied customer!" :D Now it's time to throw some of that
> > Canadian colored money at Tony's paypal account. ;)
>
> Hey! I wrote the erlang module and provided the solution to the problem
> ;)
>

ok ok. what's your paypal account? We'll give you some love too. :)
-MC
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From brian at freeswitch.org  Thu Mar  4 15:54:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 17:54:37 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
	<ec2ce17e1003041542n76c6cf6es29dd70a8c8be66a3@mail.gmail.com>
Message-ID: <E8536232-8A23-4A6D-A3C6-D4F68A28100E@freeswitch.org>

Yes, I know the muffin man. W-who lives on Drewery Lane? 

/b

On Mar 4, 2010, at 5:42 PM, jay binks wrote:

> who is the muffin man !??
> 
> yuuummm muffins... I Want some...
> fresh yummy muffins...
> 
> is BKW the muffin man ?
> 
> J
>  

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From jaybinks at gmail.com  Thu Mar  4 15:54:29 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 5 Mar 2010 09:54:29 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B90428F.1000408@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com> 
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com> 
	<4B90428F.1000408@cartissolutions.com>
Message-ID: <ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>

check this out ...
http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2

<http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2>I
think Goran Donev is Mr "Are we there yet" :)

J

On Fri, Mar 5, 2010 at 9:30 AM, Yossi Neiman <freeswitch at cartissolutions.com
> wrote:

>
> On 03/04/2010 05:15 PM, Anthony Minessale wrote:
>
> You know, that's the only question you have asked on this list.
>
> I went back and searched for your name.
>
> And you have asked it like 12 times and have been answered the same each
> time.
>
> What is the obsession with an arbitrary tag?
>
>
> Maybe we should name the 1.0.5 official release as the "Donev Release"...
>
>
>
>
> On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> No we have no deadline, we have spent countless hours preparing and we
>> still have a small list of packaging concerns.  Feel free to download the
>> code from http://latest.freeswitch.org and rest assured it *is* 1.0.5
>>
>> On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <
>> gorand at donevtechconsulting.com> wrote:
>>
>>> Howdy Freeswitch community;
>>>
>>> I was wondering if the developers are any closer to the release of 1.05
>>> version. Please don't point me to an e-mail link :) Just inquiring as to
>>> whether there is a firm close date of 1.05.
>>>
>>> Thanks.
>>> Goran
>>>
>>>
>
>
> Yossi Neiman
> Cartis Solutions, Inc. - http://www.cartissolutions.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From mayamatakeshi at gmail.com  Thu Mar  4 15:55:35 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 5 Mar 2010 08:55:35 +0900
Subject: [Freeswitch-users] How to play back a .wav file from Erlang?
In-Reply-To: <4B8FEC3F.5060102@marketelsystems.com>
References: <4B8FEC3F.5060102@marketelsystems.com>
Message-ID: <15b9404e1003041555u12978339tdc35eb6d554fe341@mail.gmail.com>

On Fri, Mar 5, 2010 at 2:22 AM, Mark Sobkow <m.sobkow at marketelsystems.com>wrote:

> I have the UUID of a parked call.  I've waited n seconds for an operator
> to answer that call, but none are available so I want to play a message
> to the customer.  How do I do that from Erlang?  I've tried the
> following, but it doesn't seem to actually play the file, and I'm not
> getting any useful information in fs_cli about the playback attempt that
> would indicate why it's not working.
>
> bgapi( eval, "uuid:1f6d7a61-fd1a-4efd-bf84-31c4eea652c2 playback
> /opt/freeswitch/sounds/en/us/callie/misc/8000/sorry.wav" )
>

Command playback is used in the dialplan.
For parked calls you must use uuid_broadcast:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
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From brian at microcomaustralia.com.au  Thu Mar  4 16:00:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Fri, 5 Mar 2010 11:00:30 +1100
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <4B903D36.2040407@cartissolutions.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
Message-ID: <3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>

On 5 March 2010 10:07, Yossi Neiman <freeswitch at cartissolutions.com> wrote:
> Standard response: ?It's in the oven baking right now. ?The muffin man
> will take it out as soon as it's ready...

Please ensure it is taken out of the oven in time and doesn't burn or
get overcooked.
-- 
Brian May <brian at microcomaustralia.com.au>



From jerry.richards at teotech.com  Thu Mar  4 16:01:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Thu, 4 Mar 2010 16:01:30 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
Message-ID: <79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>

What I see in the log where they first differ is just after FS sends the
CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
didn't work.  Do you know what PICKED_OFF means?
 
In the good case, FS logs:
 
tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities

In the bad case, FS logs:
 
tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
PICKED_OFF
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
<mailto:sofia/internal/1059 at 192.168.72.141:5060>
sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
<mailto:sofia/internal/5402 at 192.168.72.141:5060>
sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]

Best Regards,
Jerry


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Thursday, March 04, 2010 2:59 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialing **<extension>




On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry



I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC



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From anthony.minessale at gmail.com  Thu Mar  4 16:14:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Mar 2010 18:14:18 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
	<79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
Message-ID: <191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>

Did you totally ignore my instructions?
Do you want help or not?


On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  What I see in the log where they first differ is just after FS sends the
> CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
> didn't work.  Do you know what PICKED_OFF means?
>
> In the good case, FS logs:
>
> tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060(already 1)
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
> ORIGINATOR_CANCEL
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
> sleep
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
> entities
> In the bad case, FS logs:
>
> tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060(already 1)
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
> CS_REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
> CS_REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
> PICKED_OFF
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
> sleep
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
> CS_DESTROY
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
> entities
> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
> sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
> 2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
> sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
> [ORIGINATOR_CANCEL]
> Best Regards,
> Jerry
>
>  ------------------------------
> *From:* Michael Collins [mailto:msc at freeswitch.org]
> *Sent:* Thursday, March 04, 2010 2:59 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>
>
>
> On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>
>> By the way, it appears that the "**<extension>" ringing call pickup
>> feature
>> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
>> of
>> the following scenario:
>>
>> 1) 1059 calls 5381
>> 2) 5381 rings
>> 3) 5402 dials "**5381"
>>
>> In the bad case (http://pastebin.freeswitch.org/12325), all calls
>> disconnect
>> following above scenario.  In the good case
>> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>>
>> Best Regards,
>> Jerry
>>
>
> I'd suggest turning off the sofia debug stuff cuz that's just line noise
> unless you know all that stuff. You need to compare the two traces and look
> for differences. I'd start by looking at only the fs console output of each
> call and see if there's anything different between the two. See if there's a
> point where they diverge and work your way back from there. Look at the
> corresponding SIP dialogs and see if there are any clues as well. You may
> also want to capture the SIP traffic with tcpdump and analyze it in
> Wireshark which is easier on the eyes than having it buried in with the
> sofia and console logs.
>
> FWIW, I just updated to latest and I can't make this feature *NOT* work.
> I'm on 32-bit CentOS using Polycom 550 and Snom 320 phones.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Thu Mar  4 16:19:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 18:19:13 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<4B903D36.2040407@cartissolutions.com>
	<3c5cf5261003041600p23bd71betd23802e28854787e@mail.gmail.com>
Message-ID: <CD312DA2-4D85-47F0-8DCA-D0DB85453AD4@freeswitch.org>

?Find something you're passionate about and keep tremendously interested in it.?

/b

On Mar 4, 2010, at 6:00 PM, Brian May wrote:

> Please ensure it is taken out of the oven in time and doesn't burn or
> get overcooked.




From paul at apcl.us  Thu Mar  4 17:01:14 2010
From: paul at apcl.us (Paul Levin)
Date: Thu, 04 Mar 2010 20:01:14 -0500
Subject: [Freeswitch-users] how to see registration info for a
 particular user?
In-Reply-To: <4B902983.9080207@xpirio.com>
References: <4B8FCDA5.6000700@apcl.us> <4B902983.9080207@xpirio.com>
Message-ID: <4B9057DA.2010602@apcl.us>

Thanks Christian.  I thought I had tried that, but I guess I didn't.
It works.  Thanks!
     Paul


On 3/4/2010 4:43 PM, Christian L?schenkohl wrote:
> hi
>
> simply add the user id to the end of the command
> like "sofia status profile internal reg 123456"
> where 123456 is the id of the user you want to see
>
> br
>
> Paul Levin wrote:
>
>    
>> I am familiar with the console command:
>>       sofia status profile internal reg
>> to see the registration info for all currently registered users.
>>
>> Is there a modification to this command that will limit the result to
>> just one specific user registered user?
>>
>>     Thanks,
>>     Paul
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>
>    




From jcasale at activenetwerx.com  Thu Mar  4 17:09:55 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 5 Mar 2010 01:09:55 +0000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>

>check this out ...?
>http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2
>
>I think?Goran Donev is Mr "Are we there yet" :)

Lol, you guys are just fscking merciless:)

Oh and btw, How 'bout now;)

/me Ducks...



From brian at freeswitch.org  Thu Mar  4 17:14:24 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:14:24 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98C38AD@prato.activenetwerx.local>
Message-ID: <9CA09109-0FA1-4F29-A1AB-1EAB0B152EED@freeswitch.org>

How about NO.  We aren't there yet... got new batteries?

/b

On Mar 4, 2010, at 7:09 PM, Joseph L. Casale wrote:

> Oh and btw, How 'bout now;)




From infos at madovsky.org  Thu Mar  4 17:15:43 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 20:15:43 -0500
Subject: [Freeswitch-users] Release of 1.5
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
Message-ID: <B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>

Hey, this is my friend ! don't touch him !!
or I upgrade your faces with angry dog !! :D
  ----- Original Message ----- 
  From: jay binks 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 6:54 PM
  Subject: Re: [Freeswitch-users] Release of 1.5


  check this out ... 
  http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2


  I think Goran Donev is Mr "Are we there yet" :)


  J


  On Fri, Mar 5, 2010 at 9:30 AM, Yossi Neiman <freeswitch at cartissolutions.com> wrote:


    On 03/04/2010 05:15 PM, Anthony Minessale wrote: 
      You know, that's the only question you have asked on this list.

      I went back and searched for your name.

      And you have asked it like 12 times and have been answered the same each time.

      What is the obsession with an arbitrary tag?



    Maybe we should name the 1.0.5 official release as the "Donev Release"...






      On Thu, Mar 4, 2010 at 5:09 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

        No we have no deadline, we have spent countless hours preparing and we still have a small list of packaging concerns.  Feel free to download the code from http://latest.freeswitch.org and rest assured it *is* 1.0.5  



        On Thu, Mar 4, 2010 at 4:59 PM, Goran Donev <gorand at donevtechconsulting.com> wrote:

          Howdy Freeswitch community;

          I was wondering if the developers are any closer to the release of 1.05
          version. Please don't point me to an e-mail link :) Just inquiring as to
          whether there is a firm close date of 1.05.

          Thanks.
          Goran







Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com



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





  -- 
  Sincerely

  Jay



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Thu Mar  4 17:22:46 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:22:46 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
Message-ID: <8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>

I raise our angry dog... and see you a dirty dog...

/b

On Mar 4, 2010, at 7:15 PM, Madovsky wrote:

> Hey, this is my friend ! don't touch him !!
> or I upgrade your faces with angry dog !! :D
> ----- Original Message -----
> From: jay binks
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, March 04, 2010 6:54 PM
> Subject: Re: [Freeswitch-users] Release of 1.5
> 
> check this out ... 
> http://forum.sipfoundry.org/index.php?t=tree&goto=12128&S=2bbcfa5f402ed857a96b53b6944cd7d2
> 
> I think Goran Donev is Mr "Are we there yet" :)
> 
> J

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From lfurrea at gmail.com  Thu Mar  4 17:29:46 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 4 Mar 2010 19:29:46 -0600
Subject: [Freeswitch-users] event_socket outbound,
	trying to find out if a call gets put on hold 	by the phone
Message-ID: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>

Hi guys,

I am fooling (for the moment) around with event socket outbound and
basically I want to control a call that is made to a particular extension
and find out if this extension puts the call on hold. At the moment I don't
even know if the hold part is possible but I am having a hard time bridging
the call to a user.

Currently what I do is to direct the call made to user/301 to an outbound
socket, then I answer the channel and park it. Then I originate a channel to
user/301 and park it and then I try to use uuid_bridge to bridge both
channels and it seems that both channels drop.

Am I doing it correctly?

I am sure there are better ways to do it and I wanted to get some input from
you guys.

Thx a lot
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From gorand at donevtechconsulting.com  Thu Mar  4 17:34:55 2010
From: gorand at donevtechconsulting.com (Goran Donev)
Date: Thu, 4 Mar 2010 19:34:55 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 48
In-Reply-To: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <0d1e01cabc04$10896010$319c2030$@com>

I am having problems with my production server and I can not mess with
interim builds not knowing what else they can break.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
freeswitch-users-request at lists.freeswitch.org
Sent: Thursday, March 04, 2010 5:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 45, Issue 48

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific than
"Re: Contents of FreeSWITCH-users digest..."




From Russell.Mosemann at cune.org  Thu Mar  4 17:36:02 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Thu, 4 Mar 2010 19:36:02 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com><191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com><4B90428F.1000408@cartissolutions.com><ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com><B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
Message-ID: <AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>

Brian West wrote:
> I raise our angry dog... and see you a dirty dog...

OK, everyone just needs to kick back and enjoy a Mad Dog. :-)

--
Russell Mosemann




From brian at freeswitch.org  Thu Mar  4 17:42:17 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Mar 2010 19:42:17 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org><0ca201cabbee$5ee077a0$1ca166e0$@com><191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com><191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com><4B90428F.1000408@cartissolutions.com><ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com><B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
Message-ID: <33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>

I think you mean redbull.

/b

On Mar 4, 2010, at 7:36 PM, Russell Mosemann wrote:

> OK, everyone just needs to kick back and enjoy a Mad Dog. :-)




From rupa at rupa.com  Thu Mar  4 17:43:13 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 4 Mar 2010 19:43:13 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 48
In-Reply-To: <0d1e01cabc04$10896010$319c2030$@com>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
	<0d1e01cabc04$10896010$319c2030$@com>
Message-ID: <b39273531003041743s227d677fl835c3cc9787b514@mail.gmail.com>

On Thu, Mar 4, 2010 at 7:34 PM, Goran Donev
<gorand at donevtechconsulting.com>wrote:

> I am having problems with my production server and I can not mess with
> interim builds not knowing what else they can break.
>

If you are going to participate in the mailing list, please switch to
non-digest mode.

Current trunk is more stable than whatever old version you are currently
running....  Try it out in a lab machine and after verifying go for it.  No
need to wait for 1.0.5.

-- 
-Rupa
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From infos at madovsky.org  Thu Mar  4 19:55:22 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 22:55:22 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
Message-ID: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>

Is there any example of how to use pgsql as db core on wiki ?
I tried to adapt this example
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
for pgsql but no success.
I'm trying to connect to it fhru openvpn.
tcpdump -i tap1 on the destination server doesn't show any packet exchange.
I set odbc.ini as this :

[freeswitch-pgsql]
Driver   = /usr/lib64/psqlodbc.so
SERVER   = 10.0.0.1
PORT     = 5432
DATABASE = freeswitch

and set all xml conf where odbc-dsn is as this :
<param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

didn't find anything interesting on google

Thanks

Franck
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From infos at madovsky.org  Thu Mar  4 19:59:03 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 4 Mar 2010 22:59:03 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
Message-ID: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>

forgot debug :

2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
Could not connect to remote socket immedaitely

2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

thx

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 04, 2010 10:55 PM
  Subject: freeswitch and postgresql as db core


  Is there any example of how to use pgsql as db core on wiki ?
  I tried to adapt this example
  http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
  for pgsql but no success.
  I'm trying to connect to it fhru openvpn.
  tcpdump -i tap1 on the destination server doesn't show any packet exchange.
  I set odbc.ini as this :

  [freeswitch-pgsql]
  Driver   = /usr/lib64/psqlodbc.so
  SERVER   = 10.0.0.1
  PORT     = 5432
  DATABASE = freeswitch

  and set all xml conf where odbc-dsn is as this :
  <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

  didn't find anything interesting on google

  Thanks

  Franck
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From lakindia89 at gmail.com  Thu Mar  4 20:06:16 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 5 Mar 2010 09:36:16 +0530
Subject: [Freeswitch-users] event_socket outbound,
	trying to find out if a 	call gets put on hold by the phone
In-Reply-To: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>
References: <f120f4c81003041729h70223a30tb5a60206dfe1f919@mail.gmail.com>
Message-ID: <7d79b3931003042006v1720f8a8g7a0eb0b3728e1dc2@mail.gmail.com>

Can you please pastebin the dialplan! as well as the error!


On Fri, Mar 5, 2010 at 6:59 AM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi guys,
>
> I am fooling (for the moment) around with event socket outbound and
> basically I want to control a call that is made to a particular extension
> and find out if this extension puts the call on hold. At the moment I don't
> even know if the hold part is possible but I am having a hard time bridging
> the call to a user.
>
> Currently what I do is to direct the call made to user/301 to an outbound
> socket, then I answer the channel and park it. Then I originate a channel to
> user/301 and park it and then I try to use uuid_bridge to bridge both
> channels and it seems that both channels drop.
>
> Am I doing it correctly?
>
> I am sure there are better ways to do it and I wanted to get some input
> from you guys.
>
> Thx a lot
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Thu Mar  4 20:11:18 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 5 Mar 2010 09:41:18 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
Message-ID: <7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>

Ok. This is to say how the problem got solved.
Need openzap 1047 or above version.
Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz
My telco is not accepting the display IE.
Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.

Thanks for all.

On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn
<tomek.augustyn at gmail.com>wrote:

> Hello,
>
> I had similar problem and I think it is more a problem between Sangoma card
> and your E1 provider than with freeswitch.
>
> In my case it was necessary to set "origination_caller_id_number" to one
> of the telephone numbers linked to my E1 line. In other case the calls were
> rejected with "invalid information element" error.
>
> You can try Sangoma's support they are very helpful.
>
> Tomasz Augustyn
>
>
> ---------- Forwarded message ----------
> From: lakshmanan ganapathy <lakindia89 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 27 Feb 2010 10:32:08 +0530
> Subject: Re: [Freeswitch-users] smg_prid not bridging the call
> In the Dchan log it is saying Invalid Information Elements. That might be a
> problem??? But I even don't know why it is saying Invalid Information
> Element??
> Please guide me!!!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu Mar  4 21:00:20 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 00:00:20 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core [SOLVED]
Message-ID: <489E6063C33B46119F9C7B4DE9E9FB50@MOBILEE1705>

to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

odbcinst.ini

[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib64/psqlodbc.so
Setup           = /usr/lib64/libodbcpsqlS.so
FileUsage       = 1

odbc.ini

[ODBC Data Sources]
freeswitch-pgsql = ODBC for PostgreSQL

[freeswitch-pgsql]
Description = ODBC for PostgreSQL
Driver   = /usr/lib64/psqlodbc.so
Servername   = 10.0.0.1
UserName = login
Password = pwd
Port     = 5432
Database = dbdb


Thanks me ;D
  ----- Original Message ----- 
  From: Madovsky 
  To: Madovsky 
  Sent: Thursday, March 04, 2010 10:58 PM
  Subject: Re: freeswitch and postgresql as db core


  forgot debug :

  2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
  Could not connect to remote socket immedaitely

  2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

  thx

  F


    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 04, 2010 10:55 PM
    Subject: freeswitch and postgresql as db core


    Is there any example of how to use pgsql as db core on wiki ?
    I tried to adapt this example
    http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
    for pgsql but no success.
    I'm trying to connect to it fhru openvpn.
    tcpdump -i tap1 on the destination server doesn't show any packet exchange.
    I set odbc.ini as this :

    [freeswitch-pgsql]
    Driver   = /usr/lib64/psqlodbc.so
    SERVER   = 10.0.0.1
    PORT     = 5432
    DATABASE = freeswitch

    and set all xml conf where odbc-dsn is as this :
    <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

    didn't find anything interesting on google

    Thanks

    Franck
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From craig at overthewire.com.au  Thu Mar  4 21:01:43 2010
From: craig at overthewire.com.au (Craig Askings)
Date: Fri, 5 Mar 2010 15:01:43 +1000
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<0ca201cabbee$5ee077a0$1ca166e0$@com>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
	<33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
Message-ID: <8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>

Bah, Jolt is the one true caffeinated beverage.

Craig.

On 5 March 2010 11:42, Brian West <brian at freeswitch.org> wrote:

> I think you mean redbull.
>
> /b
>
> On Mar 4, 2010, at 7:36 PM, Russell Mosemann wrote:
>
> > OK, everyone just needs to kick back and enjoy a Mad Dog. :-)
>
> --
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From nagalenoj at gmail.com  Thu Mar  4 21:07:10 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Fri, 5 Mar 2010 10:37:10 +0530
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
Message-ID: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>

Dear friends,
     I've faced an issue in event socket. I would want to know why it
behaves such a way.

     My program is working with the hep of events. So, based on the received
event, the process will continue it's work. When I need DTMF, I use
recvEventTimed and in the other cases, I use recvEvent.
     So, In the mid if caller hangsup, I expect for the SERVER_DISCONNECTED
event. When the caller hangsup when I'm waiting in recvEvent, I'm getting
SERVER_DISCONNECTED. But, when I'm waiting in recvEventTimed, I'mnot
receiving SERVER_DISCONNECTED, instead receiving an undefined value.

    To handle this, I've checked esl connection inside the timeout part and
put a recvEvent, then I receive SERVER_DISCONNECTED.

    My question is, why am I not receiving SERVER_DISCONNECTED when I recv
event using recvEventTimed?!

This is a sample program I execute,

require ESL;
use IO::Socket::INET;

my $ip = "127.0.0.1";
my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '9242',
Proto => 'tcp',  Listen => 1,  Reuse => 1 );
die "Could not create socket: $!\n" unless $sock;

for(;;) {
  my $new_sock = $sock->accept();
  my $pid = fork();
  if ($pid) {
    close($new_sock);
    next;
  }

  my $host = $new_sock->sockhost();
  my $fd = fileno($new_sock);

  my $con = new ESL::ESLconnection($fd);
  my $info = $con->getInfo();
  my $uuid = $info->getHeader("unique-id");
  printf "Connected call %s, from %s\n", $uuid,
$info->getHeader("caller-caller-id-number");

  my $e = $con->filter("unique-id", $uuid);
  if ($e) {
    print $e->serialize();
  } else {
    printf("WTF?\n");
  }

  $con->events("plain", "SERVER_DISCONNECTED DTMF");
  $con->execute("answer");

  while($con->connected()) {
    $e = $con->recvEvent();   ############# CASE 1 ##
    # $e = $con->recvEventTimed(10000);       ############# CASE 2 ##

    unless ($e) {
        unless ($con->connected()) {
###############################
           $e = $con->recvEvent();                         ##
           print $e->serialize();                               ##   Added
code to get SERVER_DISCONNECTED.
           print "SERVER_DISCONNECTED";     ##
        }
###############################
        print "DTMF timeout\n";
    }
    if ($e) {
      my $name = $e->getHeader("event-name");
      print "EVENT [$name]\n";
      if ($name eq "DTMF") {
        my $digit = $e->getHeader("dtmf-digit");
        my $duration = $e->getHeader("dtmf-duration");
        print "DTMF digit $digit ($duration)\n";
      }
    }
  }
  print "BYE\n";
  close($new_sock);
}

-- 
Regards,
Nagalenoj H.
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From freeswitch.org at todandlorna.com  Thu Mar  4 21:25:22 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Thu, 04 Mar 2010 22:25:22 -0700
Subject: [Freeswitch-users] Changing Title just for Goran!
In-Reply-To: <0d1e01cabc04$10896010$319c2030$@com>
References: <mailman.26.1267746117.19150.freeswitch-users@lists.freeswitch.org>
	<0d1e01cabc04$10896010$319c2030$@com>
Message-ID: <4B9095C2.8080508@todandlorna.com>

Goran,

I can understand your desire for production stable versions.  You are 
obviously looking for stability for your client(s) and any risk you put 
them in you incur.  I have similar concerns in my own work, where 
updating FreeSWITCH could potentially cause issues that might cost the 
company money for downtime and headaches for me.  I empathize.

For a moment, please return this courtesy and try to understand this 
from the developers' perspective.  I believe you have (if you don't 
currently) done tech support for others.  Nagging didn't help you solve 
the problem any faster and often time, it makes the problem more 
difficult to approach with such a distraction.  This is effectively what 
you are doing for your needs.

Exercise some patience, or for a more timely solution, realize something 
about the FreeSWITCH project.  This is not a company with a Q-A 
department or a robust test-lab.  This is a group of individuals working 
on open-source software in various locations with various testing 
processes in place, and further help needed.  Thus, their current 
"trunk" from SVN has to be stable for their multitude of test processes, 
and code review has more relevance than Q-A.

Try trunk.  It will be just as stable for 90% of uses as 1.0.5 release 
will be.  Do a proper backup and a test run.  If you see problems, roll 
back.  It's not hard, it gets you resolved fast, and the project doesn't 
get derailed by constant questions about when it can get done.

Perhaps you'd like to give back to the project your client is relying 
on.  In that case, you can do some Q-A and test as well.  Work in Jira 
if you have such skills and do the testing.  If not, that's ok too. 
Just let other capable hands do it on their own timetable.  They're not 
lazy, and they're not unaware of the demand for 1.0.5.  It would do 
wonders if you could acknowledge that talent working for your benefit 
instead of conveying your displeasure that it is not happening on your 
specific timetable.

Cheers,

-Tod Hansmann

On 3/4/2010 6:34 PM, Goran Donev wrote:
> I am having problems with my production server and I can not mess with
> interim builds not knowing what else they can break.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> freeswitch-users-request at lists.freeswitch.org
> Sent: Thursday, March 04, 2010 5:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: FreeSWITCH-users Digest, Vol 45, Issue 48
>
> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific than
> "Re: Contents of FreeSWITCH-users digest..."
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Fri Mar  5 00:20:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:20:02 -0500
Subject: [Freeswitch-users] TEST
In-Reply-To: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>
References: <c29b930b1003040557u61972158j387c8252b52a146d@mail.gmail.com>
Message-ID: <F07A3EC7-36AB-4C5D-A965-976F8BA902D5@jerris.com>

sure, your email provider now flagged everything from this address or ip as spam, this is of course totally normal.

On Mar 4, 2010, at 8:57 AM, Milena wrote:

> you failed!!  O_O
> after the messages from clubWNC, I'm not getting any more list messages on my e-mail inbox, is it normal?




From mike at jerris.com  Fri Mar  5 00:31:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:31:02 -0500
Subject: [Freeswitch-users] about pizza demo
In-Reply-To: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
References: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
Message-ID: <1F4E6321-3C01-4994-83A0-ABE4D6BCDC75@jerris.com>

no, it would require much much more, first step is to understand how your ASR engine deals with other languages.  This is a complicated task well outside the expertise of this list and best suited to the support resources for the specific asr engine.

Mike

On Feb 26, 2010, at 5:39 AM, Madovsky wrote:

> Hi,
>  
> I'm trying to change the language of pizza demo script.
> is it need to change only words inside addItemAlias() ?
>  

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From mike at jerris.com  Fri Mar  5 00:34:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Mar 2010 03:34:35 -0500
Subject: [Freeswitch-users] Video pass problem
In-Reply-To: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
References: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
Message-ID: <FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>

double check if the behavior on this is the same with svn trunk of FreeSWITCH on something other than a bleeding edge operating system.

Mike

On Feb 25, 2010, at 7:59 AM, Mikhail Krivushin wrote:

> Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and see that we have 1280 video packets input, and only 560 passed to B leg. Anyone can point me to right direction? I can send pcap file by request.
> 
> Most time I have black screen, and then one frame can appear, and I see frozen picture pair minutes, and then other frozen picture. We have not network issues, we have good video, when bypass FS. We have not perfomance troubles to.
> 
> We have ubuntu 9.10 x64, and powerfull server board. uname:
> Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux
> 
> config:
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
> 
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From tculjaga at gmail.com  Fri Mar  5 01:44:55 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Mar 2010 10:44:55 +0100
Subject: [Freeswitch-users] Freeswitch & SNMP ( again ,
	but with a 	difference )
In-Reply-To: <ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
References: <ec2ce17e1003040038q153b5b8dgdc0137b3f152dca3@mail.gmail.com>
	<65d96fc81003041244t363450f2p8bf853c788d2295d@mail.gmail.com>
	<ec2ce17e1003041425p4e70995crf1e7b447a981e283@mail.gmail.com>
Message-ID: <65d96fc81003050144k76ffdf70rea30b109e2b19392@mail.gmail.com>

On Thu, Mar 4, 2010 at 11:25 PM, jay binks <jaybinks at gmail.com> wrote:

> HUH, well there you go..   exactly what I was after...
> someone has done exactly what I was thinking.
>
> I know you've sent it to me by email, but lets get this in the wiki.
> Ill put this up from your email, but if you have more id encourage you to
> share what youve done.
>
>

let me finish the current project ..."wholesale routing machine". and i will
put everything on wiki



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

Hi,

 

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

 

Tried the scenario as follows:

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

 

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

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

 

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

 

Thanks & Regards

Suneel


*************************************************************************
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*************************************************************************
This email and any files transmitted with it are confidential and
intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please notify
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From ahmed.ajmal at breezecom.ae  Fri Mar  5 05:02:46 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Fri, 5 Mar 2010 18:02:46 +0500
Subject: [Freeswitch-users] Losing Log lines
Message-ID: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>

Hi 

 

I am seeing the following "CRIT" log lines in my freeswitch logs:

 

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:49.633560 [CRIT]
mod_event_socket.c:161 Lost 134 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.374504 [CRIT]
mod_event_socket.c:161 Lost 218 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.893504 [CRIT]
mod_event_socket.c:161 Lost 143 log lines!

freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:59.093785 [CRIT]
mod_event_socket.c:161 Lost 27 log lines!

 

And this is happening too often now. I am relying on ESL to capture
HANGUP_COMPLETE event and insert the cdr fields in a database and it appears
from the above lines that freeswitch is missing some logs here. I am running
freeswitch on a 64-bit version of CentOS 5.4 with 4 gigs of RAM. There were
around 72 active calls(fs_cli -x "show calls count") when I noticed the
above. Also heres what my max sessions look like:

 

<param name="max-sessions" value="4000"/>

    <!--Most channels to create per second -->

<param name="sessions-per-second" value="100"/>

 

I am not sure if this is a memory issue or not but I also noticed that the
machine was running on very low memory when all this was happening. Heres
what my free -m stats looked like:

 

                       total       used       free     shared    buffers
cached

Mem:          3952       3926         26          0        186       1156

-/+ buffers/cache:        264       3688

Swap:          8189          0       8189

 

Please help me with any resolution and/or suggestions.

 

Thanks

Ahmed

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From abrownworth at bandwidth.com  Fri Mar  5 06:29:02 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Fri, 5 Mar 2010 09:29:02 -0500
Subject: [Freeswitch-users] freeswitch and postgresql as db core
In-Reply-To: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>
References: <50B10EA17E63478E9ECC55AB1FC55D71@MOBILEE1705>
Message-ID: <dbad22fd1003050629u772958eds6efa6e36385631a1@mail.gmail.com>

Try telnetting from your FreeSWITCH box to 10.0.0.1:

shell#> telnet 10.0.0.1 5432

if it doesn't connect, then this isn't a FreeSWITCH / unixODBC error. Check
your routing, etc.

If it does connect, then the error looks unixODBC related.

-a

On Thu, Mar 4, 2010 at 10:59 PM, Madovsky <infos at madovsky.org> wrote:

>  forgot debug :
>
> 2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101
> ERROR: [unixODBC]Could not connect to the server;
> Could not connect to remote socket immedaitely
>
> 2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to
> ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!
>
> thx
>
> F
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 04, 2010 10:55 PM
> *Subject:* freeswitch and postgresql as db core
>
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver   = /usr/lib64/psqlodbc.so
> SERVER   = 10.0.0.1
> PORT     = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Mar  5 06:32:55 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 08:32:55 -0600
Subject: [Freeswitch-users] Losing Log lines
In-Reply-To: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>
References: <013d01cabc64$29ae4a00$7d0ade00$@ajmal@breezecom.ae>
Message-ID: <A4B7DF27-79F3-4DB2-88DE-A19BAE249A31@freeswitch.org>

You're not reading everything off the socket so it has to drop them at some point.

/b

On Mar 5, 2010, at 7:02 AM, Ahmed Ajmal wrote:

> Hi
>  
> I am seeing the following ?CRIT? log lines in my freeswitch logs:
>  
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:49.633560 [CRIT] mod_event_socket.c:161 Lost 134 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.374504 [CRIT] mod_event_socket.c:161 Lost 218 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:54.893504 [CRIT] mod_event_socket.c:161 Lost 143 log lines!
> freeswitch.log.2010-03-05-04-30-01.1:2010-03-05 04:27:59.093785 [CRIT] mod_event_socket.c:161 Lost 27 log lines!
>  
> And this is happening too often now. I am relying on ESL to capture HANGUP_COMPLETE event and insert the cdr fields in a database and it appears from the above lines that freeswitch is missing some logs here. I am running freeswitch on a 64-bit version of CentOS 5.4 with 4 gigs of RAM. There were around 72 active calls(fs_cli -x "show calls count") when I noticed the above. Also heres what my max sessions look like:
>  
> <param name="max-sessions" value="4000"/>
>     <!--Most channels to create per second -->
> <param name="sessions-per-second" value="100"/>
>  
> I am not sure if this is a memory issue or not but I also noticed that the machine was running on very low memory when all this was happening. Heres what my free -m stats looked like:
>  
>                        total       used       free     shared    buffers     cached
> Mem:          3952       3926         26          0        186       1156
> -/+ buffers/cache:        264       3688
> Swap:          8189          0       8189
>  
> Please help me with any resolution and/or suggestions.
>  
> Thanks
> Ahmed
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dome at tel.co.th  Fri Mar  5 08:00:01 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 5 Mar 2010 23:00:01 +0700
Subject: [Freeswitch-users] freeswitch and postgresql as db core
In-Reply-To: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>
References: <2C03B3DA35D3431F89B79B2F633EDBF9@MOBILEE1705>
Message-ID: <8ccbff061003050800h12f720e9uba448422065b92fb@mail.gmail.com>

I just update wiki
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#Postgresql_8.4_on_Debian_Squeeze
It's work fine forme.

Dome C.


2010/3/5 Madovsky <infos at madovsky.org>:
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver?? = /usr/lib64/psqlodbc.so
> SERVER?? = 10.0.0.1
> PORT???? = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From m.sobkow at marketelsystems.com  Fri Mar  5 08:42:39 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 05 Mar 2010 10:42:39 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
Message-ID: <4B91347F.6070509@marketelsystems.com>


>>>         
>>     Try using park_after_bridge and catch the park event or
>>     transfer_after_bridge and set the dialplan location to transfer the call
>>     to.
>>
>>     Andrew
>>
>>       
>     What a relief.  "transfer_after_bridge" worked!  Plus it provided
>     an obvious way for me to collect the call result code, fire it
>     back to Erlang, and leave the operator waiting for the next
>     incoming call. :)
>
> "Another satisfied customer!" :D Now it's time to throw some of that 
> Canadian colored money at Tony's paypal account. ;)
> BTW, what kind of solution are you developing?
> -MC
Tech support call center solution right now, with bidirectional 
calling.  If a customer calls in and no one is available, they have the 
option of requesting a callback instead of waiting online, so both 
inbound and outbound calls have to be handled.  I'm also having fun with 
skill-based routing of calls (which is more complex than mod_fifo can 
handle.)



From infos at madovsky.org  Fri Mar  5 08:45:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 11:45:04 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db core [SOLVED]
Message-ID: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>

I sent An email last night but didn't work,
so I retry it now...


to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

odbcinst.ini

[PostgreSQL]
Description     = ODBC for PostgreSQL
Driver          = /usr/lib64/psqlodbc.so
Setup           = /usr/lib64/libodbcpsqlS.so
FileUsage       = 1

odbc.ini

[ODBC Data Sources]
freeswitch-pgsql = ODBC for PostgreSQL

[freeswitch-pgsql]
Description = ODBC for PostgreSQL
Driver   = /usr/lib64/psqlodbc.so
Servername   = 10.0.0.1
UserName = login
Password = pwd
Port     = 5432
Database = dbdb


Thanks me ;D
  ----- Original Message ----- 
  From: Madovsky 
  To: Madovsky 
  Sent: Thursday, March 04, 2010 10:58 PM
  Subject: Re: freeswitch and postgresql as db core


  forgot debug :

  2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
  Could not connect to remote socket immedaitely

  2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

  thx

  F


    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 04, 2010 10:55 PM
    Subject: freeswitch and postgresql as db core


    Is there any example of how to use pgsql as db core on wiki ?
    I tried to adapt this example
    http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
    for pgsql but no success.
    I'm trying to connect to it fhru openvpn.
    tcpdump -i tap1 on the destination server doesn't show any packet exchange.
    I set odbc.ini as this :

    [freeswitch-pgsql]
    Driver   = /usr/lib64/psqlodbc.so
    SERVER   = 10.0.0.1
    PORT     = 5432
    DATABASE = freeswitch

    and set all xml conf where odbc-dsn is as this :
    <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

    didn't find anything interesting on google

    Thanks

    Franck
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From anthony.minessale at gmail.com  Fri Mar  5 08:45:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 10:45:34 -0600
Subject: [Freeswitch-users] Release of 1.5
In-Reply-To: <8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>
References: <mailman.18.1267741609.19150.freeswitch-users@lists.freeswitch.org>
	<191c3a031003041509q388996adq1810a4212752ef71@mail.gmail.com>
	<191c3a031003041515g5f85e535ib27e0546ee483eda@mail.gmail.com>
	<4B90428F.1000408@cartissolutions.com>
	<ec2ce17e1003041554n2c531fa6vd4d5009f9fbefd8b@mail.gmail.com>
	<B32AB946D87D4EE988E1FCA342E388F1@MOBILEE1705>
	<8782F8E0-F079-4BD4-9F7A-CBBA4B66026A@freeswitch.org>
	<AF9DA5FE8AA24E64BDD5EC1E80F87551@cune.pri>
	<33E87EE7-5179-4C0D-8B00-AFCC0209898F@freeswitch.org>
	<8cc991dd1003042101h30508b9eg1d63fd9c2c33d608@mail.gmail.com>
Message-ID: <191c3a031003050845u1fb2380bg6d0638379d2d29ae@mail.gmail.com>

No mad dog is the preferred beverage of winos across the land!

On Mar 4, 2010 11:07 PM, "Craig Askings" <craig at overthewire.com.au> wrote:

Bah, Jolt is the one true caffeinated beverage.

Craig.



On 5 March 2010 11:42, Brian West <brian at freeswitch.org> wrote:
>
> I think you mean redbull.
>
> ...
-- 
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From m.sobkow at marketelsystems.com  Fri Mar  5 08:45:42 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 05 Mar 2010 10:45:42 -0600
Subject: [Freeswitch-users] Play message upon call anwser
In-Reply-To: <87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
References: <BLU110-DS256DEB3BD9142EC15687FA73B0@phx.gbl>
	<87f2f3b91003022003x8de4c92v42620a48d5bf4d78@mail.gmail.com>
Message-ID: <4B913536.70407@marketelsystems.com>

Michael Collins wrote:
>
>
> On Tue, Mar 2, 2010 at 1:15 PM, Todd <phunk0000 at hotmail.com 
> <mailto:phunk0000 at hotmail.com>> wrote:
>
>     Hey List..  I am trying to setup my dial plan so whenever a call
>     is answered a recording is played to the person who answered the
>     phone before the call is connected?.  Any help would be great.
>       Thanks
>
> Are you talking about the group confirm feature?
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> -MC
You might also try uuid_broadcast if you're doing code-driven 
dialplans.  That's what I ended up using for my Erlang-Freeswitch 
integration.



From infos at madovsky.org  Fri Mar  5 08:53:55 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 11:53:55 -0500
Subject: [Freeswitch-users] about pizza demo
References: <1871231B5C344CB08333565ED2EC2260@MOBILEE1705>
	<1F4E6321-3C01-4994-83A0-ABE4D6BCDC75@jerris.com>
Message-ID: <796886F978564D60A19E8B282E810127@MOBILEE1705>

Mike,

I took time to learn more about this feature,
I understand now how it works. If I have time
I will try to create an ASR engine for French and Italian

Regards

  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 3:31 AM
  Subject: Re: [Freeswitch-users] about pizza demo


  no, it would require much much more, first step is to understand how your ASR engine deals with other languages.  This is a complicated task well outside the expertise of this list and best suited to the support resources for the specific asr engine.


  Mike


  On Feb 26, 2010, at 5:39 AM, Madovsky wrote:


    Hi,

    I'm trying to change the language of pizza demo script.
    is it need to change only words inside addItemAlias() ?





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From abrownworth at bandwidth.com  Fri Mar  5 08:54:22 2010
From: abrownworth at bandwidth.com (Anders Brownworth)
Date: Fri, 5 Mar 2010 11:54:22 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db core
	[SOLVED]
In-Reply-To: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
Message-ID: <dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>

I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity
through your tap1 device. Does this work:

telnet 10.0.0.1 5432

-Anders

On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

>  I sent An email last night but didn't work,
> so I retry it now...
>
> to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10
> update rpm and Postgresql source)
>
> odbcinst.ini
>
> [PostgreSQL]
> Description     = ODBC for PostgreSQL
> Driver          = /usr/lib64/psqlodbc.so
> Setup           = /usr/lib64/libodbcpsqlS.so
> FileUsage       = 1
> odbc.ini
>
> [ODBC Data Sources]
> freeswitch-pgsql = ODBC for PostgreSQL
>
> [freeswitch-pgsql]
> Description = ODBC for PostgreSQL
> Driver   = /usr/lib64/psqlodbc.so
> Servername   = 10.0.0.1
> UserName = login
> Password = pwd
> Port     = 5432
> Database = dbdb
>
> Thanks me ;D
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* Madovsky <infos at madovsky.org>
> *Sent:* Thursday, March 04, 2010 10:58 PM
> *Subject:* Re: freeswitch and postgresql as db core
>
> forgot debug :
>
> 2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101
> ERROR: [unixODBC]Could not connect to the server;
> Could not connect to remote socket immedaitely
>
> 2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to
> ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!
>
> thx
>
> F
>
>  ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 04, 2010 10:55 PM
> *Subject:* freeswitch and postgresql as db core
>
> Is there any example of how to use pgsql as db core on wiki ?
> I tried to adapt this example
> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
> for pgsql but no success.
> I'm trying to connect to it fhru openvpn.
> tcpdump -i tap1 on the destination server doesn't show any packet exchange.
> I set odbc.ini as this :
>
> [freeswitch-pgsql]
> Driver   = /usr/lib64/psqlodbc.so
> SERVER   = 10.0.0.1
> PORT     = 5432
> DATABASE = freeswitch
> and set all xml conf where odbc-dsn is as this :
> <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />
>
> didn't find anything interesting on google
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri Mar  5 09:06:39 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 12:06:39 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db
	core[SOLVED]
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
	<dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>
Message-ID: <D0C806F2267A419A92D60E140DCD57BE@MOBILEE1705>

No, it works without problem
  ----- Original Message ----- 
  From: Anders Brownworth 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 11:54 AM
  Subject: Re: [Freeswitch-users] Fw: freeswitch and postgresql as db core[SOLVED]


  I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity through your tap1 device. Does this work:

  telnet 10.0.0.1 5432

  -Anders


  On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

    I sent An email last night but didn't work,
    so I retry it now...


    to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

    odbcinst.ini

    [PostgreSQL]
    Description     = ODBC for PostgreSQL
    Driver          = /usr/lib64/psqlodbc.so
    Setup           = /usr/lib64/libodbcpsqlS.so
    FileUsage       = 1

    odbc.ini

    [ODBC Data Sources]
    freeswitch-pgsql = ODBC for PostgreSQL

    [freeswitch-pgsql]
    Description = ODBC for PostgreSQL
    Driver   = /usr/lib64/psqlodbc.so
    Servername   = 10.0.0.1
    UserName = login
    Password = pwd
    Port     = 5432
    Database = dbdb


    Thanks me ;D
      ----- Original Message ----- 
      From: Madovsky 
      To: Madovsky 
      Sent: Thursday, March 04, 2010 10:58 PM
      Subject: Re: freeswitch and postgresql as db core


      forgot debug :

      2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
      Could not connect to remote socket immedaitely

      2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

      thx

      F


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 04, 2010 10:55 PM
        Subject: freeswitch and postgresql as db core


        Is there any example of how to use pgsql as db core on wiki ?
        I tried to adapt this example
        http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
        for pgsql but no success.
        I'm trying to connect to it fhru openvpn.
        tcpdump -i tap1 on the destination server doesn't show any packet exchange.
        I set odbc.ini as this :

        [freeswitch-pgsql]
        Driver   = /usr/lib64/psqlodbc.so
        SERVER   = 10.0.0.1
        PORT     = 5432
        DATABASE = freeswitch

        and set all xml conf where odbc-dsn is as this :
        <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

        didn't find anything interesting on google

        Thanks

        Franck

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






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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Fri Mar  5 09:07:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 5 Mar 2010 12:07:04 -0500
Subject: [Freeswitch-users] Fw: freeswitch and postgresql as db
	core[SOLVED]
References: <E55EEC0B9C07485C91E272AAD2D145B1@MOBILEE1705>
	<dbad22fd1003050854s37dc7bf5p4de000f3eff69bf3@mail.gmail.com>
Message-ID: <6AA8ACBF2F5C45E68DFA552625F8FD29@MOBILEE1705>

I wrote below, it solved
  ----- Original Message ----- 
  From: Anders Brownworth 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 05, 2010 11:54 AM
  Subject: Re: [Freeswitch-users] Fw: freeswitch and postgresql as db core[SOLVED]


  I suspect your problem is not FreeSWITCH or unixODBC but rather connectivity through your tap1 device. Does this work:

  telnet 10.0.0.1 5432

  -Anders


  On Fri, Mar 5, 2010 at 11:45 AM, Madovsky <infos at madovsky.org> wrote:

    I sent An email last night but didn't work,
    so I retry it now...


    to make PostgreSQL 8.4.2 works with ODBC 2.2.12 (tested from fedora 10 update rpm and Postgresql source)

    odbcinst.ini

    [PostgreSQL]
    Description     = ODBC for PostgreSQL
    Driver          = /usr/lib64/psqlodbc.so
    Setup           = /usr/lib64/libodbcpsqlS.so
    FileUsage       = 1

    odbc.ini

    [ODBC Data Sources]
    freeswitch-pgsql = ODBC for PostgreSQL

    [freeswitch-pgsql]
    Description = ODBC for PostgreSQL
    Driver   = /usr/lib64/psqlodbc.so
    Servername   = 10.0.0.1
    UserName = login
    Password = pwd
    Port     = 5432
    Database = dbdb


    Thanks me ;D
      ----- Original Message ----- 
      From: Madovsky 
      To: Madovsky 
      Sent: Thursday, March 04, 2010 10:58 PM
      Subject: Re: freeswitch and postgresql as db core


      forgot debug :

      2010-03-04 22:42:33.174950 [ERR] switch_odbc.c:313 STATE: 08001 CODE 101 ERROR: [unixODBC]Could not connect to the server;
      Could not connect to remote socket immedaitely

      2010-03-04 22:42:33.174967 [CRIT] mod_nibblebill.c:210 Cannot connect to ODBC driver/database freeswitch-pgsql (user: login / pass pwd)!

      thx

      F


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 04, 2010 10:55 PM
        Subject: freeswitch and postgresql as db core


        Is there any example of how to use pgsql as db core on wiki ?
        I tried to adapt this example
        http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
        for pgsql but no success.
        I'm trying to connect to it fhru openvpn.
        tcpdump -i tap1 on the destination server doesn't show any packet exchange.
        I set odbc.ini as this :

        [freeswitch-pgsql]
        Driver   = /usr/lib64/psqlodbc.so
        SERVER   = 10.0.0.1
        PORT     = 5432
        DATABASE = freeswitch

        and set all xml conf where odbc-dsn is as this :
        <param name="core-db-dsn" value="freeswitch-pgsql:login:pwd" />

        didn't find anything interesting on google

        Thanks

        Franck

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






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


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From dome at tel.co.th  Fri Mar  5 09:14:39 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 00:14:39 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912) Problem.
Message-ID: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>

Dear All,
        I found big odbc problem when i try to move to FS 1.0.5.
in same server i'm testing 1.0.4 and 1.0.5 and same config
1. config  limit.conf.xml use odbc (i use postgresql)
 <param name="odbc-dsn" value="tuxza:admin:xxx"/>
2. create dialplan in context public for test limit
  <condition field="destination_number" expression="^(1111)$">
       <action application="limit" data="$${domain} 0838833133 500"/>
      <action application="answer"/>
      <action application="sleep" data="60000"/>
</condition>
3. use sipp generate test call
   sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
xx.xx.xx.xx:5080

1.0.4 work fine i found 2 connection in postgresql server
but 1.0.5 use 1 connection for 1 calls. and get error when call over
100 calls. (postgresql maximum client connect is 100 )

So i'm not sure its' BUG or not ?


BG

Dome C.



From msc at freeswitch.org  Fri Mar  5 09:24:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Mar 2010 09:24:52 -0800
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <4B91347F.6070509@marketelsystems.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<4B91347F.6070509@marketelsystems.com>
Message-ID: <87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>

> > "Another satisfied customer!" :D Now it's time to throw some of that
> > Canadian colored money at Tony's paypal account. ;)
> > BTW, what kind of solution are you developing?
> > -MC
> Tech support call center solution right now, with bidirectional
> calling.  If a customer calls in and no one is available, they have the
> option of requesting a callback instead of waiting online, so both
> inbound and outbound calls have to be handled.  I'm also having fun with
> skill-based routing of calls (which is more complex than mod_fifo can
> handle.)
>

Very interesting. Okay, keep us posted on how your setup is coming along.
You might also want to join the Wednesday morning conference calls and talk
to some of the other users as well as listen to some of the presentations.
Next week Math will be discussing mod_sofia in excruciating detail. :)

-MC
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From brian at freeswitch.org  Fri Mar  5 09:32:30 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 11:32:30 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>
References: <4B8FFDA7.50708@marketelsystems.com>
	<20100304184755.GK1751@hijacked.us>
	<4B9035B4.6030901@marketelsystems.com>
	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<4B91347F.6070509@marketelsystems.com>
	<87f2f3b91003050924j4efe2088x868c550239b87bb6@mail.gmail.com>
Message-ID: <F3130139-91DF-491A-ACAB-8C1976A49422@freeswitch.org>

Better clear your schedule for a few days for that talk :P

/b

On Mar 5, 2010, at 11:24 AM, Michael Collins wrote:

> 
> Very interesting. Okay, keep us posted on how your setup is coming along. You might also want to join the Wednesday morning conference calls and talk to some of the other users as well as listen to some of the presentations. Next week Math will be discussing mod_sofia in excruciating detail. :)
> 
> -MC




From edpimentl at gmail.com  Fri Mar  5 10:16:55 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 5 Mar 2010 13:16:55 -0500
Subject: [Freeswitch-users] smart-http
Message-ID: <9dc4a1671003051016l11955a62wb2dde816cf3b1d13@mail.gmail.com>

This is just an FYI and get your thinking on the value of smart-http for
future use.
http://progit.org/2010/03/04/smart-http.html

-E
http://vCardCloud.com
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From mike at van.lammeren.net  Fri Mar  5 11:04:54 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:04:54 -0500
Subject: [Freeswitch-users] Which Lua script am I?
Message-ID: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>

Hello!

I am working on a high-availability system and need a Lua script that can
run in the background and loop forever, periodically waking up to accomplish
some tasks. I would like to be able to handle at least 10 concurrent tasks.

My first idea was to have a simple background script that would call another
script, let's call it the 'task' script, every 5 seconds or so. This would
spread the work around, and if any task script quit with an error, the
background script would remain running, and would continue to launch new
task scripts.

However, I can see no way for the background script to launch other Lua
scripts. Am I missing something? I need something like session:execute(),
but unfortunately this background script has no session, and I'm not sure if
session:execute() can even make a call like the "luarun" console command.

So, my next ideas was to have 10 copies of the background script. Each
script would loop forever. Upon launch, each script would perform a single
task, then sleep for 60 seconds. This allows handling of 10 concurrent
tasks.

However, there are some problems with running 10 copies of the same script,
which I believe can be solved if each script could be given a unique
identifier. One problem is that I would like the scripts to actually sleep
for a variable amount of time, say between 55 and 65 seconds, so that the
workload would spread around. Another, similar problem, is that each script
needs to 'claim' a number of database records from a table that is queuing
up work for the scripts. Once again, if each script had a unique identifier,
then I could use an update query to effectively 'set and test' a record that
would be tied to that script instance.

I could use a random number to identify each script, but since math.random
is seeded with a timestamp that only resolves to seconds, then all 10
scripts will be given the same sequence of random numbers, and all 10 will
run in lock-step with each other. So, random numbers are out. (If I had a
unique ID to seed the random number generator, then I could just use that
unique ID and wouldn't need random numbers.)

Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
launches at startup?

Or is there a better way of handling this situation altogether? I'm open to
any suggestions that people would care to make!

(I would entertain the idea of a single script that would start with
FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
but that brings me back to the original problem of not knowing how to launch
other Lua scripts under FreeSWITCH.)

Mike van Lammeren
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From msc at freeswitch.org  Fri Mar  5 11:16:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Mar 2010 11:16:26 -0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>

May I ask what the end goal is? It could be that there are better tools in
the toolbox to help you...

-MC

On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I am working on a high-availability system and need a Lua script that can
> run in the background and loop forever, periodically waking up to accomplish
> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>
> My first idea was to have a simple background script that would call
> another script, let's call it the 'task' script, every 5 seconds or so. This
> would spread the work around, and if any task script quit with an error, the
> background script would remain running, and would continue to launch new
> task scripts.
>
> However, I can see no way for the background script to launch other Lua
> scripts. Am I missing something? I need something like session:execute(),
> but unfortunately this background script has no session, and I'm not sure if
> session:execute() can even make a call like the "luarun" console command.
>
> So, my next ideas was to have 10 copies of the background script. Each
> script would loop forever. Upon launch, each script would perform a single
> task, then sleep for 60 seconds. This allows handling of 10 concurrent
> tasks.
>
> However, there are some problems with running 10 copies of the same script,
> which I believe can be solved if each script could be given a unique
> identifier. One problem is that I would like the scripts to actually sleep
> for a variable amount of time, say between 55 and 65 seconds, so that the
> workload would spread around. Another, similar problem, is that each script
> needs to 'claim' a number of database records from a table that is queuing
> up work for the scripts. Once again, if each script had a unique identifier,
> then I could use an update query to effectively 'set and test' a record that
> would be tied to that script instance.
>
> I could use a random number to identify each script, but since math.random
> is seeded with a timestamp that only resolves to seconds, then all 10
> scripts will be given the same sequence of random numbers, and all 10 will
> run in lock-step with each other. So, random numbers are out. (If I had a
> unique ID to seed the random number generator, then I could just use that
> unique ID and wouldn't need random numbers.)
>
> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
> launches at startup?
>
> Or is there a better way of handling this situation altogether? I'm open to
> any suggestions that people would care to make!
>
> (I would entertain the idea of a single script that would start with
> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
> but that brings me back to the original problem of not knowing how to launch
> other Lua scripts under FreeSWITCH.)
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Mar  5 11:16:39 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 13:16:39 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <7C4D5C83-9A67-4024-BC68-4B92353545CD@freeswitch.org>

What exactly are your scripts doing?

/b

On Mar 5, 2010, at 1:04 PM, Mike van Lammeren wrote:

> Or is there a better way of handling this situation altogether? I'm open to any suggestions that people would care to make!




From mike at van.lammeren.net  Fri Mar  5 11:32:39 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:32:39 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
Message-ID: <5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>

I can only talk about the project in vague terms, due to client
confidentiality, but here goes:

Many thousands of devices will be reporting events, via HTTP Post, to a
central server farm. There will be at least 5 or 6 web servers recording
these events, and a similar number of VoIP servers running FreeSWITCH.
Communication between the web servers and VoIP servers is via a shared MySQL
Cluster. Certain events will result in FreeSWITCH originating a phone call,
playing a voice message to a human, then recording a number typed in by that
human.

It is important that each event result in just one phone call. We don't want
people bothered with multiple, simultaneous calls, and we don't want things
to be missed, either.

What do you think?


On Fri, Mar 5, 2010 at 2:16 PM, Michael Collins <msc at freeswitch.org> wrote:

> May I ask what the end goal is? It could be that there are better tools in
> the toolbox to help you...
>
> -MC
>
> On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net>wrote:
>
>> Hello!
>>
>> I am working on a high-availability system and need a Lua script that can
>> run in the background and loop forever, periodically waking up to accomplish
>> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>>
>> My first idea was to have a simple background script that would call
>> another script, let's call it the 'task' script, every 5 seconds or so. This
>> would spread the work around, and if any task script quit with an error, the
>> background script would remain running, and would continue to launch new
>> task scripts.
>>
>> However, I can see no way for the background script to launch other Lua
>> scripts. Am I missing something? I need something like session:execute(),
>> but unfortunately this background script has no session, and I'm not sure if
>> session:execute() can even make a call like the "luarun" console command.
>>
>> So, my next ideas was to have 10 copies of the background script. Each
>> script would loop forever. Upon launch, each script would perform a single
>> task, then sleep for 60 seconds. This allows handling of 10 concurrent
>> tasks.
>>
>> However, there are some problems with running 10 copies of the same
>> script, which I believe can be solved if each script could be given a unique
>> identifier. One problem is that I would like the scripts to actually sleep
>> for a variable amount of time, say between 55 and 65 seconds, so that the
>> workload would spread around. Another, similar problem, is that each script
>> needs to 'claim' a number of database records from a table that is queuing
>> up work for the scripts. Once again, if each script had a unique identifier,
>> then I could use an update query to effectively 'set and test' a record that
>> would be tied to that script instance.
>>
>> I could use a random number to identify each script, but since math.random
>> is seeded with a timestamp that only resolves to seconds, then all 10
>> scripts will be given the same sequence of random numbers, and all 10 will
>> run in lock-step with each other. So, random numbers are out. (If I had a
>> unique ID to seed the random number generator, then I could just use that
>> unique ID and wouldn't need random numbers.)
>>
>> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
>> launches at startup?
>>
>> Or is there a better way of handling this situation altogether? I'm open
>> to any suggestions that people would care to make!
>>
>> (I would entertain the idea of a single script that would start with
>> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
>> but that brings me back to the original problem of not knowing how to launch
>> other Lua scripts under FreeSWITCH.)
>>
>> Mike van Lammeren
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Mar  5 11:36:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 13:36:25 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <191c3a031003051136n36f6aed1m4adfdcd925ffaa71@mail.gmail.com>

indeed you are missing something.

The FSAPI interface, see the API object in the lua docs and the "luarun" api
command.

You will probably find this knowledge very useful but even more important
you may want to know that high-availability and script do not go together as
well as high-availability and C, for instance.


On Fri, Mar 5, 2010 at 1:04 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Hello!
>
> I am working on a high-availability system and need a Lua script that can
> run in the background and loop forever, periodically waking up to accomplish
> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>
> My first idea was to have a simple background script that would call
> another script, let's call it the 'task' script, every 5 seconds or so. This
> would spread the work around, and if any task script quit with an error, the
> background script would remain running, and would continue to launch new
> task scripts.
>
> However, I can see no way for the background script to launch other Lua
> scripts. Am I missing something? I need something like session:execute(),
> but unfortunately this background script has no session, and I'm not sure if
> session:execute() can even make a call like the "luarun" console command.
>
> So, my next ideas was to have 10 copies of the background script. Each
> script would loop forever. Upon launch, each script would perform a single
> task, then sleep for 60 seconds. This allows handling of 10 concurrent
> tasks.
>
> However, there are some problems with running 10 copies of the same script,
> which I believe can be solved if each script could be given a unique
> identifier. One problem is that I would like the scripts to actually sleep
> for a variable amount of time, say between 55 and 65 seconds, so that the
> workload would spread around. Another, similar problem, is that each script
> needs to 'claim' a number of database records from a table that is queuing
> up work for the scripts. Once again, if each script had a unique identifier,
> then I could use an update query to effectively 'set and test' a record that
> would be tied to that script instance.
>
> I could use a random number to identify each script, but since math.random
> is seeded with a timestamp that only resolves to seconds, then all 10
> scripts will be given the same sequence of random numbers, and all 10 will
> run in lock-step with each other. So, random numbers are out. (If I had a
> unique ID to seed the random number generator, then I could just use that
> unique ID and wouldn't need random numbers.)
>
> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
> launches at startup?
>
> Or is there a better way of handling this situation altogether? I'm open to
> any suggestions that people would care to make!
>
> (I would entertain the idea of a single script that would start with
> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
> but that brings me back to the original problem of not knowing how to launch
> other Lua scripts under FreeSWITCH.)
>
> Mike van Lammeren
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at van.lammeren.net  Fri Mar  5 11:38:37 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:38:37 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
Message-ID: <5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>

So much for theory!

Take a look at this sample script.

-- math.randomseed(os.time())

for i=1,9 do -- final version will loop forever
    freeswitch.consoleLog("info", "wake-up #" .. i .. ", random number: " ..
math.random(111,999) .. "\n")
    -- Sleep for a while
    freeswitch.msleep(5000)
end

freeswitch.consoleLog("warning", "Script end.\n")

return 0


If I launch 10 copies of this Lua script with FreeSWITCH, with the
'startup-script' param in lua.conf.xml, then it turns out that each time
through the loop for each script does actually produce a unique random
number. If I comment out the first line, thus seeding the random number
generator, then all 10 scripts give the same 'random' number the first time
through the loop, then a unique random number for each subsequent loop!

Apparently, I don't really have the problem I thought I had.

All the same, I'm still open to suggestions on better ways to achieve my
ultimate goals!

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:32 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> I can only talk about the project in vague terms, due to client
> confidentiality, but here goes:
>
> Many thousands of devices will be reporting events, via HTTP Post, to a
> central server farm. There will be at least 5 or 6 web servers recording
> these events, and a similar number of VoIP servers running FreeSWITCH.
> Communication between the web servers and VoIP servers is via a shared MySQL
> Cluster. Certain events will result in FreeSWITCH originating a phone call,
> playing a voice message to a human, then recording a number typed in by that
> human.
>
> It is important that each event result in just one phone call. We don't
> want people bothered with multiple, simultaneous calls, and we don't want
> things to be missed, either.
>
> What do you think?
>
>
> On Fri, Mar 5, 2010 at 2:16 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> May I ask what the end goal is? It could be that there are better tools in
>> the toolbox to help you...
>>
>> -MC
>>
>> On Fri, Mar 5, 2010 at 11:04 AM, Mike van Lammeren <mike at van.lammeren.net
>> > wrote:
>>
>>> Hello!
>>>
>>> I am working on a high-availability system and need a Lua script that can
>>> run in the background and loop forever, periodically waking up to accomplish
>>> some tasks. I would like to be able to handle at least 10 concurrent tasks.
>>>
>>> My first idea was to have a simple background script that would call
>>> another script, let's call it the 'task' script, every 5 seconds or so. This
>>> would spread the work around, and if any task script quit with an error, the
>>> background script would remain running, and would continue to launch new
>>> task scripts.
>>>
>>> However, I can see no way for the background script to launch other Lua
>>> scripts. Am I missing something? I need something like session:execute(),
>>> but unfortunately this background script has no session, and I'm not sure if
>>> session:execute() can even make a call like the "luarun" console command.
>>>
>>> So, my next ideas was to have 10 copies of the background script. Each
>>> script would loop forever. Upon launch, each script would perform a single
>>> task, then sleep for 60 seconds. This allows handling of 10 concurrent
>>> tasks.
>>>
>>> However, there are some problems with running 10 copies of the same
>>> script, which I believe can be solved if each script could be given a unique
>>> identifier. One problem is that I would like the scripts to actually sleep
>>> for a variable amount of time, say between 55 and 65 seconds, so that the
>>> workload would spread around. Another, similar problem, is that each script
>>> needs to 'claim' a number of database records from a table that is queuing
>>> up work for the scripts. Once again, if each script had a unique identifier,
>>> then I could use an update query to effectively 'set and test' a record that
>>> would be tied to that script instance.
>>>
>>> I could use a random number to identify each script, but since
>>> math.random is seeded with a timestamp that only resolves to seconds, then
>>> all 10 scripts will be given the same sequence of random numbers, and all 10
>>> will run in lock-step with each other. So, random numbers are out. (If I had
>>> a unique ID to seed the random number generator, then I could just use that
>>> unique ID and wouldn't need random numbers.)
>>>
>>> Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH
>>> launches at startup?
>>>
>>> Or is there a better way of handling this situation altogether? I'm open
>>> to any suggestions that people would care to make!
>>>
>>> (I would entertain the idea of a single script that would start with
>>> FreeSWITCH, then launch 10 other scripts, giving each a unique identifier,
>>> but that brings me back to the original problem of not knowing how to launch
>>> other Lua scripts under FreeSWITCH.)
>>>
>>> Mike van Lammeren
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From dave at 3c.co.uk  Fri Mar  5 11:40:14 2010
From: dave at 3c.co.uk (David Knell)
Date: Fri, 5 Mar 2010 12:40:14 -0700
Subject: [Freeswitch-users] Which Lua script am I?
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
Message-ID: <F14E9C432BF0489FB02210622CF267F9@DELL9>

Hi Mike,

You can spawn a new script with something like

api = freeswitch.API();
reply = api:executeString("luarun /tmp/msg.lua");

- and you can pass parameters, accessed in the normal way.

So there's no reason why you couldn't have a background script running which spawns one worker per task that needs doing; each worker can do its task and then exit.

Cheers --

Dave
  ----- Original Message ----- 
  From: Mike van Lammeren 
  To: FreeSWITCH Users 
  Sent: Friday, March 05, 2010 12:04 PM
  Subject: [Freeswitch-users] Which Lua script am I?


  Hello!


  I am working on a high-availability system and need a Lua script that can run in the background and loop forever, periodically waking up to accomplish some tasks. I would like to be able to handle at least 10 concurrent tasks.


  My first idea was to have a simple background script that would call another script, let's call it the 'task' script, every 5 seconds or so. This would spread the work around, and if any task script quit with an error, the background script would remain running, and would continue to launch new task scripts.


  However, I can see no way for the background script to launch other Lua scripts. Am I missing something? I need something like session:execute(), but unfortunately this background script has no session, and I'm not sure if session:execute() can even make a call like the "luarun" console command.


  So, my next ideas was to have 10 copies of the background script. Each script would loop forever. Upon launch, each script would perform a single task, then sleep for 60 seconds. This allows handling of 10 concurrent tasks.


  However, there are some problems with running 10 copies of the same script, which I believe can be solved if each script could be given a unique identifier. One problem is that I would like the scripts to actually sleep for a variable amount of time, say between 55 and 65 seconds, so that the workload would spread around. Another, similar problem, is that each script needs to 'claim' a number of database records from a table that is queuing up work for the scripts. Once again, if each script had a unique identifier, then I could use an update query to effectively 'set and test' a record that would be tied to that script instance.


  I could use a random number to identify each script, but since math.random is seeded with a timestamp that only resolves to seconds, then all 10 scripts will be given the same sequence of random numbers, and all 10 will run in lock-step with each other. So, random numbers are out. (If I had a unique ID to seed the random number generator, then I could just use that unique ID and wouldn't need random numbers.)


  Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH launches at startup?


  Or is there a better way of handling this situation altogether? I'm open to any suggestions that people would care to make!


  (I would entertain the idea of a single script that would start with FreeSWITCH, then launch 10 other scripts, giving each a unique identifier, but that brings me back to the original problem of not knowing how to launch other Lua scripts under FreeSWITCH.)


  Mike van Lammeren




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Fri Mar  5 11:42:54 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 13:42:54 -0600
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
Message-ID: <5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>

Write it in C and use the FreeSWITCH task scheduler API.

/b

On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:

> All the same, I'm still open to suggestions on better ways to achieve my ultimate goals!




From mike at van.lammeren.net  Fri Mar  5 11:59:48 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 14:59:48 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
Message-ID: <5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>

Thanks so much, everyone! Awesome advice, as usual!

Due to a deadline situation, I'm going with the Lua script idea. However, if
it gives us any trouble after the looming deadline, then I will definitely
go the C route.

Thanks again!

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:

> Write it in C and use the FreeSWITCH task scheduler API.
>
> /b
>
> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>
> > All the same, I'm still open to suggestions on better ways to achieve my
> ultimate goals!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at van.lammeren.net  Fri Mar  5 12:17:35 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Fri, 5 Mar 2010 15:17:35 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
Message-ID: <5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>

In case anyone's interested, I just found something cool. You can pass
parameters to the Lua script from the lua.conf.xml file, like so:

    <param name="startup-script" value="temp.lua script1"/>
    <param name="startup-script" value="temp.lua script2"/>
    <param name="startup-script" value="temp.lua script3"/>

I can see that being useful.

Mike van Lammeren


On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net>wrote:

> Thanks so much, everyone! Awesome advice, as usual!
>
> Due to a deadline situation, I'm going with the Lua script idea. However,
> if it gives us any trouble after the looming deadline, then I will
> definitely go the C route.
>
> Thanks again!
>
> Mike van Lammeren
>
>
> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Write it in C and use the FreeSWITCH task scheduler API.
>>
>> /b
>>
>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>>
>> > All the same, I'm still open to suggestions on better ways to achieve my
>> ultimate goals!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From rob4manhere at gmail.com  Fri Mar  5 12:31:35 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 14:31:35 -0600
Subject: [Freeswitch-users] ESL Event
Message-ID: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>

Hi all,

I'm sending a custom event via an outbound ESL socket (using ESL's  
pymod ESL.py and _ESL.so).

Does this look right?

         event = ESL.ESLevent("CUSTOM", "test::event1")
         event.addHeader("Name1", "Value1")
         self.esl.sendEvent(event)


It used to work (r16593) but now doesn't on trunk (r16897).  I have a  
lot more information which I'll submit in jira, but just wanted to  
check that I'm not missing something obvious about sending custom  
events before I do.

All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3 (stock).

Rob

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From brian at freeswitch.org  Fri Mar  5 12:39:56 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 5 Mar 2010 14:39:56 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
Message-ID: <828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>

Wouldn't that be event.sendEvent?

/b

On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:

> 
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)

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From anthony.minessale at gmail.com  Fri Mar  5 12:43:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 14:43:26 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
Message-ID: <191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>

are you trying to send the event to a specific channel or fire the event?

The default action is fire the event, if you want to send it to a specific
channel you need to
add the unique-id header with the appropriate uuid.



On Fri, Mar 5, 2010 at 2:31 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> Hi all,
>
> I'm sending a custom event via an outbound ESL socket (using ESL's pymod
> ESL.py and _ESL.so).
>
> Does this look right?
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> It used to work (r16593) but now doesn't on trunk (r16897).  I have a lot
> more information which I'll submit in jira, but just wanted to check that
> I'm not missing something obvious about sending custom events before I do.
>
> All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3 (stock).
>
> Rob
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rob4manhere at gmail.com  Fri Mar  5 12:49:40 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 14:49:40 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<191c3a031003051243p5b9d36efle46774208b44aa0d@mail.gmail.com>
Message-ID: <26BCAA97-D813-4C84-AEA2-3149EDB3CE84@gmail.com>

Sorry I didn't specify.

I'm trying to fire an event, which is picked up by another event  
listening daemon.


On Mar 5, 2010, at 2:43 PM, Anthony Minessale wrote:

> are you trying to send the event to a specific channel or fire the  
> event?
>
> The default action is fire the event, if you want to send it to a  
> specific channel you need to
> add the unique-id header with the appropriate uuid.
>
>
>
> On Fri, Mar 5, 2010 at 2:31 PM, Rob Forman <rob4manhere at gmail.com>  
> wrote:
> Hi all,
>
> I'm sending a custom event via an outbound ESL socket (using ESL's  
> pymod ESL.py and _ESL.so).
>
> Does this look right?
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> It used to work (r16593) but now doesn't on trunk (r16897).  I have  
> a lot more information which I'll submit in jira, but just wanted to  
> check that I'm not missing something obvious about sending custom  
> events before I do.
>
> All done on trunk release, 64-bit CentOS 5.3, and python 2.4.3  
> (stock).
>
> Rob
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rob4manhere at gmail.com  Fri Mar  5 14:02:30 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 16:02:30 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
Message-ID: <926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>

I don't think so.  I see sendEvent under the ESLconnection class in  
ESL.py.  And it used to work.

Just doing a sanity check.  If it looks right I'll submit a jira ticket.



On Mar 5, 2010, at 2:39 PM, Brian West wrote:

> Wouldn't that be event.sendEvent?
>
> /b
>
> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>
>>
>>         event = ESL.ESLevent("CUSTOM", "test::event1")
>>         event.addHeader("Name1", "Value1")
>>         self.esl.sendEvent(event)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Fri Mar  5 14:57:23 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Mar 2010 14:57:23 -0800
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com><87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com><79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
	<191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
Message-ID: <15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>

Sorry, I just looked for diffs in the logs I had.  Thank you for all of your
good advice.  I do appreciate it!  I did try a snom 360 phone (but not a
polycomm) and it behaved the same, however, I did find that one of my
servers (with a 2-day newer FS version and different config) does not have
this problem.  So I will isolate the issue between my config or just upgrade
to the newer FS.
 
Thank You and Best Regards,
Jerry
 



  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, March 04, 2010 4:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialing **<extension>


Did you totally ignore my instructions?
Do you want help or not?



On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


What I see in the log where they first differ is just after FS sends the
CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
didn't work.  Do you know what PICKED_OFF means?
 
In the good case, FS logs:
 
tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities

In the bad case, FS logs:
 
tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060
(already 1)
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
CS_REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
PICKED_OFF
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to sleep
2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
CS_DESTROY
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
(sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
entities
2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
<mailto:sofia/internal/1059 at 192.168.72.141:5060>
sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
<mailto:sofia/internal/5402 at 192.168.72.141:5060>
sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
[ORIGINATOR_CANCEL]

Best Regards,
Jerry


  _____  

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Thursday, March 04, 2010 2:59 PM 

To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Dialing **<extension>




On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:



By the way, it appears that the "**<extension>" ringing call pickup feature
is intermittent.  Sometimes it works, sometimes not.  I posted two traces of
the following scenario:

1) 1059 calls 5381
2) 5381 rings
3) 5402 dials "**5381"

In the bad case (http://pastebin.freeswitch.org/12325), all calls disconnect
following above scenario.  In the good case
(http://pastebin.freeswitch.org/12326), 5402 connects to 1059.

Best Regards,
Jerry



I'd suggest turning off the sofia debug stuff cuz that's just line noise
unless you know all that stuff. You need to compare the two traces and look
for differences. I'd start by looking at only the fs console output of each
call and see if there's anything different between the two. See if there's a
point where they diverge and work your way back from there. Look at the
corresponding SIP dialogs and see if there are any clues as well. You may
also want to capture the SIP traffic with tcpdump and analyze it in
Wireshark which is easier on the eyes than having it buried in with the
sofia and console logs.

FWIW, I just updated to latest and I can't make this feature *NOT* work. I'm
on 32-bit CentOS using Polycom 550 and Snom 320 phones.

-MC




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






-- 
Anthony Minessale II

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

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

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


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From anthony.minessale at gmail.com  Fri Mar  5 15:11:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 17:11:41 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
	<926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
Message-ID: <191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>

found your problem
fixed in r16921

I think in the future maybe just open the jira right away, it's easy to
close them as "not a bug" if you are wrong.
If you are right it spares us the trouble of tracking issues over email.


On Fri, Mar 5, 2010 at 4:02 PM, Rob Forman <rob4manhere at gmail.com> wrote:

> I don't think so.  I see sendEvent under the ESLconnection class in ESL.py.
>  And it used to work.
>
> Just doing a sanity check.  If it looks right I'll submit a jira ticket.
>
>
>
> On Mar 5, 2010, at 2:39 PM, Brian West wrote:
>
> Wouldn't that be event.sendEvent?
>
> /b
>
> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>
>
>         event = ESL.ESLevent("CUSTOM", "test::event1")
>         event.addHeader("Name1", "Value1")
>         self.esl.sendEvent(event)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar  5 15:15:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 17:15:06 -0600
Subject: [Freeswitch-users] Dialing **<extension>
In-Reply-To: <15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>
References: <A4199FB0C9B5435899E1797CC581D21B@greyhawk.tonecommander.com>
	<87f2f3b91003041458p3f5463cav844283daa6ab02db@mail.gmail.com>
	<79FDBF1AB958496AAFB277D5614EFD4B@greyhawk.tonecommander.com>
	<191c3a031003041614j458e6af6sf49df61fd5194cd0@mail.gmail.com>
	<15FDA7562D044B4DBCD7654C73F13F07@greyhawk.tonecommander.com>
Message-ID: <191c3a031003051515u232820a3ob8b13ab70377549c@mail.gmail.com>

What I am trying to tell you going forward is to open issues on jira not the
mailing list I have to say it 4-10 times a day to people.  We cant follow
issues over email, we get literally 100+ emails a day some days.  It's for
your own good.  We need proper workflow....



On Fri, Mar 5, 2010 at 4:57 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  Sorry, I just looked for diffs in the logs I had.  Thank you for all of
> your good advice.  I do appreciate it!  I did try a snom 360 phone (but
> not a polycomm) and it behaved the same, however, I did find that one of my
> servers (with a 2-day newer FS version and different config) does not have
> this problem.  So I will isolate the issue between my config or just upgrade
> to the newer FS.
>
> Thank You and Best Regards,
> Jerry
>
>
>  ------------------------------
> *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com]
> *Sent:* Thursday, March 04, 2010 4:14 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>
> Did you totally ignore my instructions?
> Do you want help or not?
>
>
> On Thu, Mar 4, 2010 at 6:01 PM, Jerry Richards <jerry.richards at teotech.com
> > wrote:
>
>>  What I see in the log where they first differ is just after FS sends the
>> CANCEL to the original ringing phone.  FS logs a cause: PICKED_OFF when it
>> didn't work.  Do you know what PICKED_OFF means?
>>
>> In the good case, FS logs:
>>
>> tport_pend(0x88348e8): pending 0x8942cc0 for udp/192.168.72.141:5060(already 1)
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
>> ORIGINATOR_CANCEL
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
>> sleep
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:13:04.772196 [DEBUG] switch_core_session.c:1161 Session 177
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
>> entities
>> In the bad case, FS logs:
>>
>> tport_pend(0x88348e8): pending 0xb6dd6468 for udp/192.168.72.141:5060(already 1)
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:333
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Running State Change
>> CS_REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/sip:5381 at 192.168.72.58:5060 Standard REPORTING, cause:
>> PICKED_OFF
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:585
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State REPORTING going to
>> sleep
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_state_machine.c:327
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/sip:5381 at 192.168.72.58:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1161 Session 199
>> (sofia/internal/sip:5381 at 192.168.72.58:5060) Locked, Waiting on external
>> entities
>> 2010-03-04 12:33:45.001563 [DEBUG] switch_core_session.c:1019 Send signal
>> sofia/internal/1059 at 192.168.72.141:5060 [BREAK]
>> 2010-03-04 12:33:45.001563 [NOTICE] switch_ivr_bridge.c:740 Hangup
>> sofia/internal/5402 at 192.168.72.141:5060 [CS_SOFT_EXECUTE]
>> [ORIGINATOR_CANCEL]
>> Best Regards,
>> Jerry
>>
>>  ------------------------------
>> *From:* Michael Collins [mailto:msc at freeswitch.org]
>> *Sent:* Thursday, March 04, 2010 2:59 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Dialing **<extension>
>>
>>
>>
>> On Thu, Mar 4, 2010 at 2:14 PM, Jerry Richards <
>> jerry.richards at teotech.com> wrote:
>>
>>>
>>> By the way, it appears that the "**<extension>" ringing call pickup
>>> feature
>>> is intermittent.  Sometimes it works, sometimes not.  I posted two traces
>>> of
>>> the following scenario:
>>>
>>> 1) 1059 calls 5381
>>> 2) 5381 rings
>>> 3) 5402 dials "**5381"
>>>
>>> In the bad case (http://pastebin.freeswitch.org/12325), all calls
>>> disconnect
>>> following above scenario.  In the good case
>>> (http://pastebin.freeswitch.org/12326), 5402 connects to 1059.
>>>
>>> Best Regards,
>>> Jerry
>>>
>>
>> I'd suggest turning off the sofia debug stuff cuz that's just line noise
>> unless you know all that stuff. You need to compare the two traces and look
>> for differences. I'd start by looking at only the fs console output of each
>> call and see if there's anything different between the two. See if there's a
>> point where they diverge and work your way back from there. Look at the
>> corresponding SIP dialogs and see if there are any clues as well. You may
>> also want to capture the SIP traffic with tcpdump and analyze it in
>> Wireshark which is easier on the eyes than having it buried in with the
>> sofia and console logs.
>>
>> FWIW, I just updated to latest and I can't make this feature *NOT* work.
>> I'm on 32-bit CentOS using Polycom 550 and Snom 320 phones.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rupa at rupa.com  Fri Mar  5 15:29:49 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Mar 2010 17:29:49 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
Message-ID: <b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>

hmm.. at some point postgres is falling behind and rather than waiting,
we're opening another connection.  <ponder/>  can you open a jira on this?
 we probably need to implement connection limits in the core db connection
thingie.

On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>        I found big odbc problem when i try to move to FS 1.0.5.
> in same server i'm testing 1.0.4 and 1.0.5 and same config
> 1. config  limit.conf.xml use odbc (i use postgresql)
>  <param name="odbc-dsn" value="tuxza:admin:xxx"/>
> 2. create dialplan in context public for test limit
>  <condition field="destination_number" expression="^(1111)$">
>       <action application="limit" data="$${domain} 0838833133 500"/>
>      <action application="answer"/>
>      <action application="sleep" data="60000"/>
> </condition>
> 3. use sipp generate test call
>   sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
> xx.xx.xx.xx:5080
>
> 1.0.4 work fine i found 2 connection in postgresql server
> but 1.0.5 use 1 connection for 1 calls. and get error when call over
> 100 calls. (postgresql maximum client connect is 100 )
>
> So i'm not sure its' BUG or not ?
>
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From lloyd.aloysius at gmail.com  Fri Mar  5 15:34:32 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 5 Mar 2010 18:34:32 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
Message-ID: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>

Hi All,

I follow all the steps in the wiki.

Multi-tenant work properly , I need to setup DNS SRV and SIP for each
Domain.

If I do not have the access the domains DNS how can I make Multi-tenant
work.

Thank you.
Lloyd
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From jerry.richards at teotech.com  Fri Mar  5 15:36:08 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 5 Mar 2010 15:36:08 -0800
Subject: [Freeswitch-users] 415 Unsupported Media Handling
In-Reply-To: <191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>
References: <AE66B63BAF424BBCBE585E244E0C42B4@greyhawk.tonecommander.com><835A36B2-1C04-42B0-86C1-36528E93589D@freeswitch.org><ADCCADE853D6427CBC7CD978E7E0A0ED@greyhawk.tonecommander.com>
	<191c3a031002261255x7e037e6ftc119eab0ed21b69b@mail.gmail.com>
Message-ID: <BD3EC285AA0B4887B173E8AF99051693@greyhawk.tonecommander.com>

Sorry again for the late reply on this issue.  I think I understand what
you're saying.  Since IMs are not sent in the context of a dialog (it's
simply a send/receive handshake), the b2bua has nothing to tell the
originator.  I will investigate the possibility of getting the Bria phones
to send plain text.  Regarding commercial support options for FS, I raised
the issue to management here, but I'm not sure what the outcome of that will
be (it's not my decision to make).
 
Thanks And Best Regards,
Jerry
 


  _____  

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Friday, February 26, 2010 12:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling


It says SHOULD, not MUST right?

The message passing in FS is abstracted and protocol agnostic and we are a
b2bua not a proxy in terms of SIP.
You are sending a message to 1 UA on FS who is accepting the message and
delivering it to the core who is happy to receive messages in any format.
Then it's routed back out another sip dialog where it's rejected.  It's too
late to go tell the sender that is unacceptable because we already happily
accepted it (messages are not always passed out to other phones they can
easily be directed at other internal resources).  We don't know what the
content-type means as we are a neutral party in the whole thing so there is
not much else we can do but violate this scope issue and break out of our
role as a neutral party and translate it to plain text and try again which
is not very elegant.

If FS was a proxy software, like openser and friends, we would be passing
the data between UA in the way you expect but we are not a proxy.


Based on the frequency and specific nature of your constant inquiries, and
the likelihood that you are offering commercial VoIP services to customers.
I suggest you contact us at consulting at freeswitch.org to investigate
commercial support options for FreeSWITCH.  Even then, I am not sure I could
help you besides maybe a param to convert all text/html messages to plain
text or some other sad hack.





On Fri, Feb 26, 2010 at 1:43 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


I'm not sure I follow your comment.  The first device prefers text/html so
that's what it normally sets in the initial MESSAGE.  Devices that support
text/html will not generate this 415 error reply.  It's only devices that
don't support it that would send the 415 reply, so the issue is that the 415
is not getting passed back to the originator.

Best Regards,
Jerry



-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org]
Sent: Friday, February 26, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] 415 Unsupported Media Handling

Its really clear here you'll need to say text/plain in the content type
their accept header says they only take text/plain.

/b

On Feb 26, 2010, at 12:57 PM, Jerry Richards wrote:

>
> I have two types of devices, one supports text/html MESSAGE content
> and one that only supports text/plain MESSAGE content.  When I send an
> IM from the first to the second, the second replies with 415
> Unsupported Media Type (as shown below).  Section 8.1.3.5 of RFC 3261
> says the sender should retry using the media type acceptable to the
receiver (in this case: plain/text).
>
> The problem I have is that Freeswitch doesn't pass the error back to
> the sender (nor does it retry itself using plain/text).  So the IM is
lost.
> Does anyone see the reason why the error is not being handled correctly?
>
>
> ----------------------------------------------------------------------
> -- send 668 bytes to udp/[192.168.72.141]:5062 at 18:43:24.720446:
>   ------------------------------------------------------------------------
>   MESSAGE sip:5398 at 192.168.72.141:5062 SIP/2.0
>   Via: SIP/2.0/UDP 192.168.72.141;rport;branch=z9hG4bKXg6SNUcQa5Kre
>   Max-Forwards: 70
>   From: "5382 on 141" <sip:5382 at 192.168.72.141
<mailto:sip%3A5382 at 192.168.72.141> >;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Contact: <sip:mod_sofia at 192.168.72.141:5060>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-32M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>   Supported: timer, precondition, path, replaces
>   Content-Type: text/html
>   Content-Length: 63
>
>   <font size="2" face="Arial">hello this is Jerry from Teo</font>
>
> ----------------------------------------------------------------------
> -- recv 459 bytes from udp/[192.168.72.141]:5062 at 18:43:24.728390:
>   ------------------------------------------------------------------------
>   SIP/2.0 415 Unsupported media type
>   Via: SIP/2.0/UDP
> 192.168.72.141;rport=5060;branch=z9hG4bKXg6SNUcQa5Kre;received=192.168
> .72.14
> 1
>   From: "5382 on 141" <sip:5382 at 192.168.72.141
<mailto:sip%3A5382 at 192.168.72.141> >;tag=66661130
>   To: "5398" <sip:5398 at 192.168.72.141:5062>
>   Call-ID: a9891a15-9da9-122d-b3ad-003048d7e9f0
>   CSeq: 127444135 MESSAGE
>   Date: Fri, 26 Feb 2010 18:43:24 GMT
>   User-Agent: MobilityGateway-2.0.34078
>   Server: MobilityGateway-2.0.34078
>   Accept: text/plain
>   Content-Length: 0
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org





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





-- 
Anthony Minessale II

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

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

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


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From rob4manhere at gmail.com  Fri Mar  5 15:40:08 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Fri, 5 Mar 2010 17:40:08 -0600
Subject: [Freeswitch-users] ESL Event
In-Reply-To: <191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>
References: <4F3EF9D7-AB88-404D-A94C-E894E8DDED93@gmail.com>
	<828714BA-792C-4BB7-9AFA-039EAE629081@freeswitch.org>
	<926AA714-1B8C-480C-A67E-1D6B07ACECA5@gmail.com>
	<191c3a031003051511i5fe87c76u6fdbe28e0834169c@mail.gmail.com>
Message-ID: <64C2AEA6-E0BA-483F-BAB8-1F85BBEC070A@gmail.com>

Thanks, and will do.


On Mar 5, 2010, at 5:11 PM, Anthony Minessale wrote:

> found your problem
> fixed in r16921
>
> I think in the future maybe just open the jira right away, it's easy  
> to close them as "not a bug" if you are wrong.
> If you are right it spares us the trouble of tracking issues over  
> email.
>
>
> On Fri, Mar 5, 2010 at 4:02 PM, Rob Forman <rob4manhere at gmail.com>  
> wrote:
> I don't think so.  I see sendEvent under the ESLconnection class in  
> ESL.py.  And it used to work.
>
> Just doing a sanity check.  If it looks right I'll submit a jira  
> ticket.
>
>
>
> On Mar 5, 2010, at 2:39 PM, Brian West wrote:
>
>> Wouldn't that be event.sendEvent?
>>
>> /b
>>
>> On Mar 5, 2010, at 2:31 PM, Rob Forman wrote:
>>
>>>
>>>         event = ESL.ESLevent("CUSTOM", "test::event1")
>>>         event.addHeader("Name1", "Value1")
>>>         self.esl.sendEvent(event)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rupa at rupa.com  Fri Mar  5 15:43:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 5 Mar 2010 17:43:32 -0600
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
Message-ID: <b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>

You have access to your own DNS, so you could do something like:

companya.supersip.com
companyb.supersip.com
etc

so you are actually doing the dns in your domain (supersip.com) and
provision the phones as such.

On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I follow all the steps in the wiki.
>
> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
> Domain.
>
> If I do not have the access the domains DNS how can I make Multi-tenant
> work.
>
> Thank you.
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From anthony.minessale at gmail.com  Fri Mar  5 16:43:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Mar 2010 18:43:45 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
Message-ID: <191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>

Now you are getting a dedicated handle per thread because of our cache db
system.  Either allow a max connections that matches the most calls you
anticipate or file a bug with rupa to downgrade the mod back to sharing one
db handle globally.

On Mar 5, 2010 5:35 PM, "Rupa Schomaker" <rupa at rupa.com> wrote:

hmm.. at some point postgres is falling behind and rather than waiting,
we're opening another connection.  <ponder/>  can you open a jira on this?
 we probably need to implement connection limits in the core db connection
thingie.



On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>
> Dear All,
>        I...
-- 
-Rupa

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lloyd.aloysius at gmail.com  Fri Mar  5 17:32:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Fri, 5 Mar 2010 20:32:34 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
	<b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
Message-ID: <8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>

Thanks Rupa. I will try your suggestion.


On Fri, Mar 5, 2010 at 6:43 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> You have access to your own DNS, so you could do something like:
>
> companya.supersip.com
> companyb.supersip.com
> etc
>
> so you are actually doing the dns in your domain (supersip.com) and
> provision the phones as such.
>
> On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I follow all the steps in the wiki.
>>
>> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
>> Domain.
>>
>> If I do not have the access the domains DNS how can I make Multi-tenant
>> work.
>>
>> Thank you.
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Mar  5 19:28:41 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 6 Mar 2010 11:28:41 +0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
Message-ID: <23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>

and also you can get the script name from argv[0] if you name you
script in different names

2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
> In case anyone's interested, I just found something cool. You can pass
> parameters to the Lua script from the lua.conf.xml file, like so:
> ?? ?<param name="startup-script" value="temp.lua script1"/>
> ?? ?<param name="startup-script" value="temp.lua script2"/>
> ?? ?<param name="startup-script" value="temp.lua script3"/>
> I can see that being useful.
> Mike van Lammeren
>
> On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net>
> wrote:
>>
>> Thanks so much, everyone! Awesome advice, as usual!
>> Due to a deadline situation, I'm going with the Lua script idea. However,
>> if it gives us any trouble after the looming deadline, then I will
>> definitely go the C route.
>> Thanks again!
>> Mike van Lammeren
>>
>> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org> wrote:
>>>
>>> Write it in C and use the FreeSWITCH task scheduler API.
>>>
>>> /b
>>>
>>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>>>
>>> > All the same, I'm still open to suggestions on better ways to achieve
>>> > my ultimate goals!
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dome at tel.co.th  Fri Mar  5 19:36:07 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 10:36:07 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<191c3a031003051641u4d31ca0eg20dd5eada8ebfec0@mail.gmail.com>
	<191c3a031003051643l2cd1af05ua84bf32a47cc45d0@mail.gmail.com>
Message-ID: <8ccbff061003051936ue24c8f9pda00418a36d93313@mail.gmail.com>

How to limit cache db max connections  in FS ?

Dome C.

2010/3/6 Anthony Minessale <anthony.minessale at gmail.com>:
> Now you are getting a dedicated handle per thread because of our cache db
> system.? Either allow a max connections that matches the most calls you
> anticipate or file a bug with rupa to downgrade the mod back to sharing one
> db handle globally.
>
> On Mar 5, 2010 5:35 PM, "Rupa Schomaker" <rupa at rupa.com> wrote:
>
> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another connection. ?<ponder/> ?can you open a jira on this?
> ?we probably need to implement connection limits in the core db connection
> thingie.
>
> On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ?I...
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From dome at tel.co.th  Fri Mar  5 19:39:18 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 6 Mar 2010 10:39:18 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
Message-ID: <8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>

Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
I'll open JIRA for this case.

Dome C.

2010/3/6 Rupa Schomaker <rupa at rupa.com>:
> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another connection. ?<ponder/> ?can you open a jira on this?
> ?we probably need to implement connection limits in the core db connection
> thingie.
>
> On Fri, Mar 5, 2010 at 11:14 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ?I found big odbc problem when i try to move to FS 1.0.5.
>> in same server i'm testing 1.0.4 and 1.0.5 and same config
>> 1. config ?limit.conf.xml use odbc (i use postgresql)
>> ?<param name="odbc-dsn" value="tuxza:admin:xxx"/>
>> 2. create dialplan in context public for test limit
>> ?<condition field="destination_number" expression="^(1111)$">
>> ? ? ? <action application="limit" data="$${domain} 0838833133 500"/>
>> ? ? ?<action application="answer"/>
>> ? ? ?<action application="sleep" data="60000"/>
>> </condition>
>> 3. use sipp generate test call
>> ? sipp -rtp_echo -sn uac -r 30 -l 500 -d 30000 -s 1111 -mp 25000
>> xx.xx.xx.xx:5080
>>
>> 1.0.4 work fine i found 2 connection in postgresql server
>> but 1.0.5 use 1 connection for 1 calls. and get error when call over
>> 100 calls. (postgresql maximum client connect is 100 )
>>
>> So i'm not sure its' BUG or not ?
>>
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From infos at madovsky.org  Fri Mar  5 22:24:00 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 01:24:00 -0500
Subject: [Freeswitch-users] wiki DNS page update
Message-ID: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>

for anyone who can update this page for NAPTR ex

http://wiki.freeswitch.org/wiki/DNS


# NAPTR for SIP over UDP
example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
# NAPTR for SIP over TCP
example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
# NAPTR for SIP over TLS
example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

Regards

Franck


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

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From infos at madovsky.org  Fri Mar  5 23:28:45 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 02:28:45 -0500
Subject: [Freeswitch-users] fundamental question
Message-ID: <DD4280539BDB4AEB9FCC25ECA7BD2CF8@MOBILEE1705>

Concerning softphones not SRV compatible
how to make them work for registration with a domain name managed by NAPTR and/or SRV ?
I tried 

x-lite  --->  register
3cx    ---> not register

Thanks

Franck

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From brian at freeswitch.org  Sat Mar  6 07:42:50 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 09:42:50 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
Message-ID: <709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>

Anyone can update it thats the point.  Login and add them.

http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS

Verify them against that.  I think the flag is lowercase.

/b

On Mar 6, 2010, at 12:24 AM, Madovsky wrote:

> for anyone who can update this page for NAPTR ex
>  
> http://wiki.freeswitch.org/wiki/DNS
>  
>  
> # NAPTR for SIP over UDP
> example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
> # NAPTR for SIP over TCP
> example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
> # NAPTR for SIP over TLS
> example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.
>  
> Regards
>  
> Franck
> 
> 
> -- 
> This message has been scanned for viruses and 
> dangerous content by MailScanner, and is 
> believed to be clean. _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar  6 08:24:28 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 11:24:28 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
Message-ID: <64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>

It doesn't matter apparently,
but it needs to correct the line

# NAPTR for SIP over TLS
example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 10:42 AM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Anyone can update it thats the point.  Login and add them.


  http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


  Verify them against that.  I think the flag is lowercase.


  /b


  On Mar 6, 2010, at 12:24 AM, Madovsky wrote:


    for anyone who can update this page for NAPTR ex

    http://wiki.freeswitch.org/wiki/DNS


    # NAPTR for SIP over UDP
    example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
    # NAPTR for SIP over TCP
    example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

    Regards

    Franck



    -- 
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From infos at madovsky.org  Sat Mar  6 08:25:26 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 11:25:26 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
Message-ID: <551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>

but anyway, to be sure let's change it in lower case
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 10:42 AM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Anyone can update it thats the point.  Login and add them.


  http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


  Verify them against that.  I think the flag is lowercase.


  /b


  On Mar 6, 2010, at 12:24 AM, Madovsky wrote:


    for anyone who can update this page for NAPTR ex

    http://wiki.freeswitch.org/wiki/DNS


    # NAPTR for SIP over UDP
    example.com NAPTR 10 100 "S" "SIP+D2U" "" _sip._udp.example.com.
    # NAPTR for SIP over TCP
    example.com NAPTR 20 100 "S" "SIP+D2T" "" _sip._tcp.example.com.
    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sip._tcp.example.com.

    Regards

    Franck



    -- 
    This message has been scanned for viruses and 
    dangerous content by MailScanner, and is 
    believed to be clean. _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sat Mar  6 09:02:44 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:02:44 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
	<64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
Message-ID: <427E641C-70AA-457A-80EA-675360805515@freeswitch.org>

I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.

/b

On Mar 6, 2010, at 10:24 AM, Madovsky wrote:

> # NAPTR for SIP over TLS
> example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.
> 

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From brian at freeswitch.org  Sat Mar  6 09:09:09 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:09:09 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705>
	<709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org>
	<551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
Message-ID: <6F12E9C2-2C42-421E-89D8-139FD630D1BB@freeswitch.org>

I don't know who changed it to  _sips._tcp.domain.com. but that is 100% WRONG I want back to my zone files I used for testing and its _sip._tls and you can even google that.  

Can someone pin down something to prove me wrong?

/b

On Mar 6, 2010, at 10:25 AM, Madovsky wrote:

> but anyway, to be sure let's change it in lower case
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 10:42 AM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> Anyone can update it thats the point.  Login and add them.
> 
> http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS
> 
> Verify them against that.  I think the flag is lowercase.
> 
> /b

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From infos at madovsky.org  Sat Mar  6 09:11:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:11:05 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
Message-ID: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>

I dont' think _sip._tls is standard, but if it works, great
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:02 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.


  /b


  On Mar 6, 2010, at 10:24 AM, Madovsky wrote:


    # NAPTR for SIP over TLS
    example.com NAPTR 20 100 "S" "SIPS+D2T" "" _sips._tcp.example.com.






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From brian at freeswitch.org  Sat Mar  6 09:15:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:15:05 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
	<8D677CB395454F21B5691A27515AB558@MOBILEE1705>
Message-ID: <776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>

I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls

So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.

/b

On Mar 6, 2010, at 11:11 AM, Madovsky wrote:

> I dont' think _sip._tls is standard, but if it works, great
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:02 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.
> 
> /b

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From brian at freeswitch.org  Sat Mar  6 09:15:24 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 11:15:24 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <8D677CB395454F21B5691A27515AB558@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705>
	<427E641C-70AA-457A-80EA-675360805515@freeswitch.org>
	<8D677CB395454F21B5691A27515AB558@MOBILEE1705>
Message-ID: <30FD929F-6817-450D-A240-E4AB742D04D9@freeswitch.org>

Oh and eyebeam does _sip._tls

/b

On Mar 6, 2010, at 11:11 AM, Madovsky wrote:

> I dont' think _sip._tls is standard, but if it works, great
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:02 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.
> 
> /b

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From infos at madovsky.org  Sat Mar  6 09:35:54 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:35:54 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><551B5E333314487FB4D5AA0EDA0D9AC9@MOBILEE1705>
	<6F12E9C2-2C42-421E-89D8-139FD630D1BB@freeswitch.org>
Message-ID: <21F56D3A23CF434AA7E2844256AD0494@MOBILEE1705>

What DNS server you use ?
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:09 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I don't know who changed it to  _sips._tcp.domain.com. but that is 100% WRONG I want back to my zone files I used for testing and its _sip._tls and you can even google that.  


  Can someone pin down something to prove me wrong?


  /b


  On Mar 6, 2010, at 10:25 AM, Madovsky wrote:


    but anyway, to be sure let's change it in lower case
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 10:42 AM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      Anyone can update it thats the point.  Login and add them.


      http://wiki.freeswitch.org/wiki/Tls#Configuring_DNS_NAPTR_and_SRV_for_TLS


      Verify them against that.  I think the flag is lowercase.


      /b




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


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From infos at madovsky.org  Sat Mar  6 09:36:50 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 12:36:50 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705>
	<776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>
Message-ID: <B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>

ok
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 12:15 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls


  So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.


  /b


  On Mar 6, 2010, at 11:11 AM, Madovsky wrote:


    I dont' think _sip._tls is standard, but if it works, great
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 12:02 PM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      I'm almost 100% positive its _sip._tls. its what I used in my testing with polycoms and snoms to direct the phone to use TLS.


      /b




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From brian at freeswitch.org  Sat Mar  6 10:05:48 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 6 Mar 2010 12:05:48 -0600
Subject: [Freeswitch-users] wiki DNS page update
In-Reply-To: <B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705>
	<776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org>
	<B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
Message-ID: <46F21E0D-FDEC-49C4-9725-2B14C12CDF42@freeswitch.org>

Let clarify this ... after digging thru the code sofia looks for _sips._tcp but polycom/snom/eyebeam will look for _sip._tls, 

I use Bind.

/b

On Mar 6, 2010, at 11:36 AM, Madovsky wrote:

> ok
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, March 06, 2010 12:15 PM
> Subject: Re: [Freeswitch-users] wiki DNS page update
> 
> I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls
> 
> So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.
> 
> /b

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From anthony.minessale at gmail.com  Sat Mar  6 10:06:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 6 Mar 2010 12:06:29 -0600
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>
	<191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
Message-ID: <191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>

Did anyone read my last email?

It is only worthy of jira if its a real problem.
There is 1 db con per channel did you try increasing the max conns and
compare the performance?

On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
I'll open JIRA for this case.

Dome C.

2010/3/6 Rupa Schomaker <rupa at rupa.com>:

> hmm.. at some point postgres is falling behind and rather than waiting,
> we're opening another co...
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From infos at madovsky.org  Sat Mar  6 10:17:14 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 13:17:14 -0500
Subject: [Freeswitch-users] wiki DNS page update
References: <6219FBA750E5472A95F2467D78174484@MOBILEE1705><709C073E-42B2-4A0D-BCAC-5FAC163553FC@freeswitch.org><64963983C9224EBCA1C69CE27C37F7BE@MOBILEE1705><427E641C-70AA-457A-80EA-675360805515@freeswitch.org><8D677CB395454F21B5691A27515AB558@MOBILEE1705><776AD47A-1BDF-4EE8-87A6-7D4A9E2F1A3C@freeswitch.org><B87E6933A62D4075AF78A987E494CCF6@MOBILEE1705>
	<46F21E0D-FDEC-49C4-9725-2B14C12CDF42@freeswitch.org>
Message-ID: <0B32A78695AA4AF19FF2E286B5F29D2B@MOBILEE1705>

eyebeam doesn't use strct standard stuff
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 1:05 PM
  Subject: Re: [Freeswitch-users] wiki DNS page update


  Let clarify this ... after digging thru the code sofia looks for _sips._tcp but polycom/snom/eyebeam will look for _sip._tls, 


  I use Bind.


  /b


  On Mar 6, 2010, at 11:36 AM, Madovsky wrote:


    ok
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, March 06, 2010 12:15 PM
      Subject: Re: [Freeswitch-users] wiki DNS page update


      I know aastra does sips: but polycom/snom do sip: with transport=tls and lookup _sip._tls


      So its safe to say SIP sucks and you prob. need both but FreeSWITCH doesn't fully handle sips: because every device we had that tried it was badly busted i.e. aastra.


      /b




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


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From infos at madovsky.org  Sat Mar  6 10:19:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 6 Mar 2010 13:19:10 -0500
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk
	16912)Problem.
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com><b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com><8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com><191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
	<191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
Message-ID: <48640F1B60F64D87BCF88646668EB341@MOBILEE1705>

I use postgresql, but I can test it if you have a tools that
similate calls charge.
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 06, 2010 1:06 PM
  Subject: Re: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)Problem.


  Did anyone read my last email?

  It is only worthy of jira if its a real problem. 
  There is 1 db con per channel did you try increasing the max conns and compare the performance?



    On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:

    Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
    I'll open JIRA for this case.

    Dome C.

    2010/3/6 Rupa Schomaker <rupa at rupa.com>:

    > hmm.. at some point postgres is falling behind and rather than waiting,
    > we're opening another co...




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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dome at tel.co.th  Sat Mar  6 19:54:48 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 7 Mar 2010 10:54:48 +0700
Subject: [Freeswitch-users] ODBC connection in 1.0.5 (trunk 16912)
	Problem.
In-Reply-To: <191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
References: <8ccbff061003050914q6535dabldf8fe977255f43f6@mail.gmail.com>
	<b39273531003051529x426d71edq4786a61c832a1797@mail.gmail.com>
	<8ccbff061003051939x3949475en4b138a2066950830@mail.gmail.com>
	<191c3a031003061005s444fa099y67f5e07851fb3d8b@mail.gmail.com>
	<191c3a031003061006j5fe9a734o2f4041a2ce917d92@mail.gmail.com>
Message-ID: <8ccbff061003061954y16da64e1m16e17a743b943490@mail.gmail.com>

2010/3/7 Anthony Minessale <anthony.minessale at gmail.com>:
> Did anyone read my last email?

Sorry just read it:)

>
> It is only worthy of jira if its a real problem.
> There is 1 db con per channel did you try increasing the max conns and
> compare the performance?
max conns ? you mean in postgresql ? I think postgresql need more
resource if change max client  from 100 to 2000

Now i port mod_limit from 1.0.4 to 1.0.5 it's work for this case. but
i'm not sure about other db connect (core , sofia)
i only use mod_limit.

>
> On Mar 5, 2010 9:44 PM, "Dome Charoenyost" <dome at tel.co.th> wrote:
>
> Now i use mod_limit from 1.0.4 and fix some code for work in 1.0.5
> I'll open JIRA for this case.
>
> Dome C.
>
> 2010/3/6 Rupa Schomaker <rupa at rupa.com>:
>
>> hmm.. at some point postgres is falling behind and rather than waiting,
>> we're opening another co...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From shaheryarkh at googlemail.com  Sun Mar  7 01:38:44 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 7 Mar 2010 14:38:44 +0500
Subject: [Freeswitch-users]
	=?utf-8?q?Tienes_una_invitaci=C3=B3n_a_ClubWNC?=
In-Reply-To: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>
References: <E1Nn1sB-0001rv-PT@eclipse.websitewelcome.com>
Message-ID: <b16156851003070138x6ba2e08dred8c2d6daeb5e2e2@mail.gmail.com>

Hola,

Chicos, por favor, poner fin a este correo spam en list.I FreeSwitch estoy
haciendo unos 20-30 mensajes de este tipo todos los d?as. S?rvase no abusen
de este foro para fines publicitarios y de marketing.

Gracias.


2010/3/4 seven <dujinfang at gmail.com>

> [image: ClubWNC]<http://www.clubwnc.com/eshop/index.php?sponsor_id=0&cdr=2>
>  *Tienes una invitaci?n a ClubWNC*
>  Hola !!!
> He creado un perfil en ClubWNC donde puedo hacer listas de regalos,
> publicar rese?as sobre productos, ver a mis amigos de Facebook y lo que les
> gusta, lo que han comprado, y lo que quieren que les regalen, y quiero
> agregarte a mis amigos para que puedas verlo. Para ello, necesitas
> registrarte en ClubWNC y, despu?s, podr?s crear tambi?n tu propio perfil.
> Para registrarte en ClubWNC, sigue este enlace:
> http://www.clubwnc.com/eshop/create_eshop_account.php?cdr=2895<http://www.clubwnc.com/eshop/create_eshop_account.php?sponsor_id=0&cdr=2895>     seven
> ha invitado a freeswitch-users at lists.freeswitch.org a unirse a ClubWNC. Si
> no quieres recibir este tipo de mensajes de ClubWNC, haz clic aqu? para
> borrar tu nombre de la lista de personas suscritas.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From neil.burgess at redmatter.com  Sun Mar  7 04:31:31 2010
From: neil.burgess at redmatter.com (Neil Burgess)
Date: Sun, 7 Mar 2010 12:31:31 +0000
Subject: [Freeswitch-users] Export b-leg variables in bridge string for a
	user bridge lookup
Message-ID: <787302A89ACCE24DA8F56DA101E77C84139B8D275D@THHS2E12BE1X.hostedservice2.net>

Hi,

Wonder if anyone can help me with a bridge to user question?

We are trying to bridge to a user in the directory using a command of the form:-

session.execute("bridge","[leg_confirm=yes]user/1009");

We want to be able to pass a number of channel variables to the b-leg. In the case above, we want to pass leg_confirm to enable us to use the group_confirm functionality on the leg, but have found that whilst this is fine using something like session.execute("bridge","[leg_confirm=yes]sofia/blah/blah"), when used against a user/  lookup it doesn't set the variable.

I can see that the dial-string in the User's Directory definition would presumably need these bridge channel variables added to it somehow, but don't see how to achieve that!

We have used export in some cases to set the channel variable on the b-leg for a user which is fine, until we want to do something like session.execute("bridge","[leg_confirm=yes]user/1009, [leg_confirm=no]user/1012"), where the two b-legs have differing requirements.

Any ideas how I can achieve a working version of session.execute("bridge","[leg_confirm=yes]user/1009");

As an added twist, we are using mod_xml_curl to perform our directory lookups!

Many thanks,
Neil Burgess

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From infos at madovsky.org  Sun Mar  7 10:56:42 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 7 Mar 2010 13:56:42 -0500
Subject: [Freeswitch-users] xmpp IM
Message-ID: <660B041BADA44ADE924CD4E293B118BB@MOBILEE1705>

Is there any already existing vas that contains IM in mod_dingaling ?

Thanks

Franck
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From paul.gore.j at gmail.com  Sun Mar  7 11:03:46 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 14:03:46 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
Message-ID: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>

Greetings,
We use XML curl to fetch our dial plan extensions from a database. As per FS
wiki we can only return one XML extension upon FS request.
Now we need to return a set of extensions with same number condition and
context but with different second condition so that FS can pick up the right
one.
Is it possible at all? And if yes how response XML with multiple extensions
should look like?

Thanks a lot.
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From lloyd.aloysius at gmail.com  Sun Mar  7 11:55:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 14:55:50 -0500
Subject: [Freeswitch-users] Multi-tenant - DNS SRV and SIP Configuration
In-Reply-To: <8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>
References: <8a19bf2e1003051534k1673648fucfba348a760029fd@mail.gmail.com>
	<b39273531003051543t5e67e43bka33c3f02639906a1@mail.gmail.com>
	<8a19bf2e1003051732v3ec9e004q7217e73aed6a9726@mail.gmail.com>
Message-ID: <8a19bf2e1003071155m76bbc0kee7688929604cbcc@mail.gmail.com>

Rupa,

Thank you again. It is working great.

Lloyd


On Fri, Mar 5, 2010 at 8:32 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thanks Rupa. I will try your suggestion.
>
>
>
> On Fri, Mar 5, 2010 at 6:43 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> You have access to your own DNS, so you could do something like:
>>
>> companya.supersip.com
>> companyb.supersip.com
>> etc
>>
>> so you are actually doing the dns in your domain (supersip.com) and
>> provision the phones as such.
>>
>> On Fri, Mar 5, 2010 at 5:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>>
>>> Hi All,
>>>
>>> I follow all the steps in the wiki.
>>>
>>> Multi-tenant work properly , I need to setup DNS SRV and SIP for each
>>> Domain.
>>>
>>> If I do not have the access the domains DNS how can I make Multi-tenant
>>> work.
>>>
>>> Thank you.
>>> Lloyd
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Sun Mar  7 12:03:34 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 15:03:34 -0500
Subject: [Freeswitch-users] Voice Mail 2 Email Configuration
Message-ID: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>

Hi All,

I am running FreeSWITCH on CentOS5.4 . Sendmail already installed in the
system and running . I can send email the following way

echo "test" | mail -s testsubject lloyd.aloysius at gmail.com

Here is the FreeSWITCH configuration

*1. Extension setup*

<include>
  <user id="201">
    <params>
      <param name="password" value="201"/>
      <param name="vm-password" value="201"/>
        <param name="vm-email-all-messages" value="true"/>
      <param name="vm-attach-file" value="true" />
      <param name="vm-mailto" value="lloyd.aloysius at gmail.com"/>

    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="201"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 201"/>
      <variable name="effective_caller_id_number" value="201"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>


*2. /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml*


   <param name="mailer-app" value="sendmail"/>
    <param name="mailer-app-args" value="-t"/>
    <param name="dump-cores" value="yes"/>

But not a single voicemail deliver to email. How to troubleshoot this
problem. Am I missing any configuration?


Thanks,
Lloyd
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From nazim.agabekov at gmail.com  Sun Mar  7 12:18:07 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Mon, 08 Mar 2010 00:18:07 +0400
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
Message-ID: <4B9409FF.5060608@gmail.com>

Hello Paul

I also use mod_xml_curl for a dialplan. I've solved this issue by
generating the XML extension dynamically.
My fastcgi daemon checks the input variables like caller id, originator,
e.t.c, makes routing decision
and dynamically generates only one XML extension.
So, there is no  real need for creating complex dialplan on mod_xml_curl
output.

Nazim

On 03/07/2010 11:03 PM, paul gore wrote:
> Greetings,
> We use XML curl to fetch our dial plan extensions from a database. As
> per FS wiki we can only return one XML extension upon FS request. 
> Now we need to return a set of extensions with same number condition
> and context but with different second condition so that FS can pick up
> the right one. 
> Is it possible at all? And if yes how response XML with multiple
> extensions should look like?
>
> Thanks a lot.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From lloyd.aloysius at gmail.com  Sun Mar  7 12:36:07 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 7 Mar 2010 15:36:07 -0500
Subject: [Freeswitch-users] Voice Mail 2 Email Configuration
In-Reply-To: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>
References: <8a19bf2e1003071203u10e8919s3e659e266bfc1e19@mail.gmail.com>
Message-ID: <8a19bf2e1003071236n67f7f2d9l9c7a234ae1bb4fe2@mail.gmail.com>

Hi All,

I am answering my question. I fixed the problem it is a sendemail problem.

Thanks
Lloyd


On Sun, Mar 7, 2010 at 3:03 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I am running FreeSWITCH on CentOS5.4 . Sendmail already installed in the
> system and running . I can send email the following way
>
> echo "test" | mail -s testsubject lloyd.aloysius at gmail.com
>
> Here is the FreeSWITCH configuration
>
> *1. Extension setup*
>
> <include>
>   <user id="201">
>     <params>
>       <param name="password" value="201"/>
>       <param name="vm-password" value="201"/>
>         <param name="vm-email-all-messages" value="true"/>
>       <param name="vm-attach-file" value="true" />
>       <param name="vm-mailto" value="lloyd.aloysius at gmail.com"/>
>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="201"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 201"/>
>       <variable name="effective_caller_id_number" value="201"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
>
> *2. /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml*
>
>
>    <param name="mailer-app" value="sendmail"/>
>     <param name="mailer-app-args" value="-t"/>
>     <param name="dump-cores" value="yes"/>
>
> But not a single voicemail deliver to email. How to troubleshoot this
> problem. Am I missing any configuration?
>
>
> Thanks,
> Lloyd
>
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From m.sobkow at marketelsystems.com  Sun Mar  7 13:33:03 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sun, 07 Mar 2010 15:33:03 -0600
Subject: [Freeswitch-users] Having some trouble recording calls.
Message-ID: <4B941B8F.1070506@marketelsystems.com>

I'm using the uuid_record command to tell Freeswitch to record a call.

    RecordResult = pbx:api( uuid_record, 
Operator#pbx_operator_registry.operator_uuid ++ " start " ++ FullFileName ),

After the customer hangs up, I tell it to stop recording:

    StopRecordResult = pbx:api( uuid_record, 
Operator#pbx_operator_registry.operator_uuid ++ " stop " ++ FullFileName ),

However, I get an error "-ERR Cannot stop record session!".  Any 
subsequent attempts to start a new recording file also fail.

What am I doing wrong?



From m.sobkow at marketelsystems.com  Sun Mar  7 14:10:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sun, 07 Mar 2010 16:10:10 -0600
Subject: [Freeswitch-users] Having some trouble recording calls.
In-Reply-To: <4B941B8F.1070506@marketelsystems.com>
References: <4B941B8F.1070506@marketelsystems.com>
Message-ID: <4B942442.2050001@marketelsystems.com>

Mark Sobkow wrote:
> I'm using the uuid_record command to tell Freeswitch to record a call.
>
>    RecordResult = pbx:api( uuid_record, 
> Operator#pbx_operator_registry.operator_uuid ++ " start " ++ 
> FullFileName ),
>
> After the customer hangs up, I tell it to stop recording:
>
>    StopRecordResult = pbx:api( uuid_record, 
> Operator#pbx_operator_registry.operator_uuid ++ " stop " ++ 
> FullFileName ),
>
> However, I get an error "-ERR Cannot stop record session!".  Any 
> subsequent attempts to start a new recording file also fail.
>
> What am I doing wrong?
>
Actually the logs show a bit more.

The first time I try to record a session, I get an ok message.  When I 
stop that recording, it tells me ok as well.

But on every subsequent attempt to record part of the session, I'm 
getting errors.  It's as if Freeswitch will only allow you to record 
once for an operator session.



From paul.gore.j at gmail.com  Sun Mar  7 14:13:20 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 17:13:20 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <4B9409FF.5060608@gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
Message-ID: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>

Thanks fro the input.
I thought about such approach, the only problem I see is there that
I essentially would have to duplicate FS dial plan parsing functionality, so
my question basically is there a way to avoid that?
I have seen a Ruby dial plan example on XML curl wiki page which actually
seem to generate response with multiple extensions, but unfortunately FS
cannot read such response.

On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  Hello Paul
>
> I also use mod_xml_curl for a dialplan. I've solved this issue by
> generating the XML extension dynamically.
> My fastcgi daemon checks the input variables like caller id, originator,
> e.t.c, makes routing decision
> and dynamically generates only one XML extension.
> So, there is no  real need for creating complex dialplan on mod_xml_curl
> output.
>
> Nazim
>
>
> On 03/07/2010 11:03 PM, paul gore wrote:
>
> Greetings,
> We use XML curl to fetch our dial plan extensions from a database. As per
> FS wiki we can only return one XML extension upon FS request.
> Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?
>
>  Thanks a lot.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sun Mar  7 14:38:05 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 7 Mar 2010 14:38:05 -0800
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
Message-ID: <81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>

Can you pastebin a sample dialplan and script? I'd like to better  
visualize what problem you're trying to solve.
-MC

Sent from my iPhone

On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:

> Thanks fro the input.
> I thought about such approach, the only problem I see is there that  
> I essentially would have to duplicate FS dial plan parsing  
> functionality, so my question basically is there a way to avoid that?
> I have seen a Ruby dial plan example on XML curl wiki page which  
> actually seem to generate response with multiple extensions, but  
> unfortunately FS cannot read such response.
>
> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <nazim.agabekov at gmail.com 
> > wrote:
> Hello Paul
>
> I also use mod_xml_curl for a dialplan. I've solved this issue by  
> generating the XML extension dynamically.
> My fastcgi daemon checks the input variables like caller id,  
> originator, e.t.c, makes routing decision
> and dynamically generates only one XML extension.
> So, there is no  real need for creating complex dialplan on  
> mod_xml_curl output.
>
> Nazim
>
>
> On 03/07/2010 11:03 PM, paul gore wrote:
>> Greetings,
>> We use XML curl to fetch our dial plan extensions from a database.  
>> As per FS wiki we can only return one XML extension upon FS request.
>> Now we need to return a set of extensions with same number  
>> condition and context but with different second condition so that  
>> FS can pick up the right one.
>> Is it possible at all? And if yes how response XML with multiple  
>> extensions should look like?
>>
>> Thanks a lot.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From paul.gore.j at gmail.com  Sun Mar  7 15:07:27 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 18:07:27 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
Message-ID: <da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>

Well, basically we used to return XML dial plan response with one extension
like

<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="test9">
        <condition field="destination_number" expression="^83789$">
          <action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/>
        </condition>
      </extension></context></section></document>

and now we want to return response like that

<document type="freeswitch/xml">
  <section name="dialplan" description="RE Dial Plan For FreeSwitch">
    <context name="default">
      <extension name="test9"><condition field="username"
expression="^756755$"/><condition field="destination_number"
expression="^83789$"><action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/></condition></extension>

<extension name="test9"><condition field="username"
expression="^5564645$"/><condition field="destination_number"
expression="^83789$"><action application="bridge"
data="iax/guest at conference.freeswitch.org/888"/></condition></extension></context></section></document>


But FS seem not to understand the latter, we get "No route..."


On Sun, Mar 7, 2010 at 5:38 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Can you pastebin a sample dialplan and script? I'd like to better visualize
> what problem you're trying to solve.
> -MC
>
> Sent from my iPhone
>
> On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:
>
> Thanks fro the input.
> I thought about such approach, the only problem I see is there that
> I essentially would have to duplicate FS dial plan parsing functionality, so
> my question basically is there a way to avoid that?
> I have seen a Ruby dial plan example on XML curl wiki page which actually
> seem to generate response with multiple extensions, but unfortunately FS
> cannot read such response.
>
> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <<nazim.agabekov at gmail.com>
> nazim.agabekov at gmail.com> wrote:
>
>>  Hello Paul
>>
>> I also use mod_xml_curl for a dialplan. I've solved this issue by
>> generating the XML extension dynamically.
>> My fastcgi daemon checks the input variables like caller id, originator,
>> e.t.c, makes routing decision
>> and dynamically generates only one XML extension.
>> So, there is no  real need for creating complex dialplan on mod_xml_curl
>> output.
>>
>> Nazim
>>
>>
>> On 03/07/2010 11:03 PM, paul gore wrote:
>>
>> Greetings,
>> We use XML curl to fetch our dial plan extensions from a database. As per
>> FS wiki we can only return one XML extension upon FS request.
>> Now we need to return a set of extensions with same number condition and
>> context but with different second condition so that FS can pick up the right
>> one.
>> Is it possible at all? And if yes how response XML with multiple
>> extensions should look like?
>>
>>  Thanks a lot.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From paul.gore.j at gmail.com  Sun Mar  7 16:19:31 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 19:19:31 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<81F54B71-2C10-448D-8053-6E151E8CCB3D@freeswitch.org>
	<da4ab26e1003071507k3acda66bv76d32e27d2f3e22b@mail.gmail.com>
Message-ID: <da4ab26e1003071619w4851481ct1c322e98ae099e0d@mail.gmail.com>

Actually to be correct - FS does read such response, but looks like it
processes only the first one from the top, and if its condition evaluates to
false it does not process further extensions.

On Sun, Mar 7, 2010 at 6:07 PM, paul gore <paul.gore.j at gmail.com> wrote:

> Well, basically we used to return XML dial plan response with one extension
> like
>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>     <context name="default">
>       <extension name="test9">
>         <condition field="destination_number" expression="^83789$">
>           <action application="bridge" data="iax/guest at conference.freeswitch.org/888"/>
>         </condition>
>       </extension></context></section></document>
>
> and now we want to return response like that
>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>     <context name="default">
>       <extension name="test9"><condition field="username" expression="^756755$"/><condition field="destination_number" expression="^83789$"><action application="bridge" data="iax/guest at conference.freeswitch.org/888"/></condition></extension>
>
> <extension name="test9"><condition field="username" expression="^5564645$"/><condition field="destination_number" expression="^83789$"><action application="bridge" data="iax/guest at conference.freeswitch.org/888"/></condition></extension></context></section></document>
>
>
> But FS seem not to understand the latter, we get "No route..."
>
>
> On Sun, Mar 7, 2010 at 5:38 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Can you pastebin a sample dialplan and script? I'd like to better
>> visualize what problem you're trying to solve.
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 7, 2010, at 2:13 PM, paul gore <paul.gore.j at gmail.com> wrote:
>>
>> Thanks fro the input.
>> I thought about such approach, the only problem I see is there that
>> I essentially would have to duplicate FS dial plan parsing functionality, so
>> my question basically is there a way to avoid that?
>> I have seen a Ruby dial plan example on XML curl wiki page which actually
>> seem to generate response with multiple extensions, but unfortunately FS
>> cannot read such response.
>>
>> On Sun, Mar 7, 2010 at 3:18 PM, Nazim Agabekov <<nazim.agabekov at gmail.com>
>> nazim.agabekov at gmail.com> wrote:
>>
>>>  Hello Paul
>>>
>>> I also use mod_xml_curl for a dialplan. I've solved this issue by
>>> generating the XML extension dynamically.
>>> My fastcgi daemon checks the input variables like caller id, originator,
>>> e.t.c, makes routing decision
>>> and dynamically generates only one XML extension.
>>> So, there is no  real need for creating complex dialplan on mod_xml_curl
>>> output.
>>>
>>> Nazim
>>>
>>>
>>> On 03/07/2010 11:03 PM, paul gore wrote:
>>>
>>> Greetings,
>>> We use XML curl to fetch our dial plan extensions from a database. As per
>>> FS wiki we can only return one XML extension upon FS request.
>>> Now we need to return a set of extensions with same number condition and
>>> context but with different second condition so that FS can pick up the right
>>> one.
>>> Is it possible at all? And if yes how response XML with multiple
>>> extensions should look like?
>>>
>>>  Thanks a lot.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From josh at radianttiger.com  Sun Mar  7 16:28:31 2010
From: josh at radianttiger.com (Josh Rivers)
Date: Sun, 7 Mar 2010 16:28:31 -0800
Subject: [Freeswitch-users] Conference Recording
Message-ID: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>

On last week's conference call, someone was recording, did that recording
get put online somewhere?

Josh
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From brian at freeswitch.org  Sun Mar  7 16:32:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 18:32:28 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
Message-ID: <5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>

You generate exactly ONE extension which was the exact way it was designed to be used.  You don't have to parse anything only look at the variables you care about and run with it.  Its nothing too complex or overly intensive to do.  

/b

On Mar 7, 2010, at 4:13 PM, paul gore wrote:

> Thanks fro the input.
> I thought about such approach, the only problem I see is there that I essentially would have to duplicate FS dial plan parsing functionality, so my question basically is there a way to avoid that? 
> I have seen a Ruby dial plan example on XML curl wiki page which actually seem to generate response with multiple extensions, but unfortunately FS cannot read such response.




From paul.gore.j at gmail.com  Sun Mar  7 17:30:02 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 20:30:02 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
Message-ID: <da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>

OK, so that means multiple extensions not possible.
Yes, it's not overly intensive, but nevertheless I actually have to parse
regex in the second condition to find out which extension matches incoming
number and I wanted instead to let FS do that for me.
It is a bit complex since we store complete extension XML in DB, so I have
to retrieve all the extensions matching first condition, extract their
second condition, then run them in loop... Looks like I can be better off
using JS for that instead.
Thank you everybody.

On Sun, Mar 7, 2010 at 7:32 PM, Brian West <brian at freeswitch.org> wrote:

> You generate exactly ONE extension which was the exact way it was designed
> to be used.  You don't have to parse anything only look at the variables you
> care about and run with it.  Its nothing too complex or overly intensive to
> do.
>
> /b
>
> On Mar 7, 2010, at 4:13 PM, paul gore wrote:
>
> > Thanks fro the input.
> > I thought about such approach, the only problem I see is there that I
> essentially would have to duplicate FS dial plan parsing functionality, so
> my question basically is there a way to avoid that?
> > I have seen a Ruby dial plan example on XML curl wiki page which actually
> seem to generate response with multiple extensions, but unfortunately FS
> cannot read such response.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From djbinter at gmail.com  Sun Mar  7 17:40:45 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Sun, 7 Mar 2010 17:40:45 -0800
Subject: [Freeswitch-users] mod xml rpc
Message-ID: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>

I just enabled mod xml rpc to test the web interface to retrieve the
voicemail; however, every time when I tried to access via
http://(fs_ip):8080/api/voicemail/web<http://%28fs_ip%29:8080/api/voicemail/web>,
somehow FS process got killed by itself.
I wonder whether anyone experienced this problem. I am running r16928.

Thank you,
Dorn B.
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From brian at freeswitch.org  Sun Mar  7 17:46:33 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 19:46:33 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
Message-ID: <84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>

Apparently you haven't used a SQL db before.  SELECT * from extensions where did='1234';  You can query for EXACTLY the info you need.

/b

On Mar 7, 2010, at 7:30 PM, paul gore wrote:

> It is a bit complex since we store complete extension XML in DB, so I have to retrieve all the extensions matching first condition, extract their second condition, then run them in loop... Looks like I can be better off using JS for that instead.




From brian at freeswitch.org  Sun Mar  7 17:47:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 19:47:12 -0600
Subject: [Freeswitch-users] mod xml rpc
In-Reply-To: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>
References: <94f7dfb11003071740ld55ca1dhb3a7e24af7be027c@mail.gmail.com>
Message-ID: <63AEAD12-F18F-4E61-B753-5F285ED067BB@freeswitch.org>

Please read this http://wiki.freeswitch.org/wiki/Reporting_Bugs and report a bug to jira.

/b

On Mar 7, 2010, at 7:40 PM, DJB INTERNATIONAL wrote:

> I just enabled mod xml rpc to test the web interface to retrieve the voicemail; however, every time when I tried to access via http://(fs_ip):8080/api/voicemail/web, somehow FS process got killed by itself.  
> I wonder whether anyone experienced this problem.  I am running r16928.
> 
> 
> Thank you,
> Dorn B.

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From paul.gore.j at gmail.com  Sun Mar  7 18:03:14 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Sun, 7 Mar 2010 21:03:14 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
Message-ID: <da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>

:-))
 I used it a lot. Unfortunately we do not store did number as a field, as I
said we store complete extension XML, with extension number and context as a
key.
So essentially I will retrieve a bunch of extension XMLs which I need to
parse and run against regex matcher to find out the one, which is what
exactly FS is doing when XML dial plan is defined in a static file.



On Sun, Mar 7, 2010 at 8:46 PM, Brian West <brian at freeswitch.org> wrote:

> Apparently you haven't used a SQL db before.  SELECT * from extensions
> where did='1234';  You can query for EXACTLY the info you need.
>
> /b
>
> On Mar 7, 2010, at 7:30 PM, paul gore wrote:
>
> > It is a bit complex since we store complete extension XML in DB, so I
> have to retrieve all the extensions matching first condition, extract their
> second condition, then run them in loop... Looks like I can be better off
> using JS for that instead.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Mar  7 18:07:59 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 7 Mar 2010 21:07:59 -0500
Subject: [Freeswitch-users] alias in profile
Message-ID: <16B4AB8FDE0149A98BFA5128C312957B@MOBILEE1705>

Is anyone can enlight me of the definition
of "alias" in profile and for what is it made for ?

Thanks

Franck
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From brian at freeswitch.org  Sun Mar  7 18:17:19 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 7 Mar 2010 20:17:19 -0600
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
Message-ID: <A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>

Sounds like you zigged where you should have zagged.

/b

On Mar 7, 2010, at 8:03 PM, paul gore wrote:

> :-))
>  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.




From rob4manhere at gmail.com  Sun Mar  7 18:34:00 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Sun, 7 Mar 2010 20:34:00 -0600
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
Message-ID: <e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>

Nice to see so many people using it in so many different ways.  For those
using ec2 or xen, have you noticed this in your logs?

2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating timer,
please wait...
2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer resolution of
4000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may experience
audio problems.


"You may experience audio problems."  Seems pretty clear.  Can that just be
ignored?

Rob


On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
> wrote:

> I used to do tests on conferencing using linode.com, which is Xen VMs
> without issues, although there were not much load on it.
>
> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
>> pretty well. Of course no much load.
>>
>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
>> wrote:
>> >>  >  Cheers, Chris.
>> >>  >  --
>> >>  >  RightScale, Inc.
>> >>
>> >> I doubt they'll be switching any time soon.  I think his employer IS
>> >> RightScale :)
>> >
>> > Oooops, I'll learn to pay attention to signatures :)
>> >
>> >
>> >>
>> >>
>> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>> >>
>> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>> >>>> RightScale's management dashboard"?
>> >>>>
>> >>>> There are some significant differences to the configuration when
>> >>>> using RightScale vs. bare bones AWS; and given this is on my
>> >>>> employer's time I need to keep it relevant...
>> >>>
>> >>> Chris,
>> >>> I know can be more work, but the best would be to let know the
>> >>> differences too, and how it would be using bare bone AWS. That's also
>> >>> good for your employer, if he decide to switch back from RightScale...
>> >>> :)
>> >>>
>> >>> -giovanni
>> >>>
>> >>>
>> >>>>
>> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>> >>>> a public AMI (you don't need to be a RightScale customer to use it)
>> >>>>
>> >>>> Cheers, Chris.
>> >>>>
>> >>>> -----Original Message-----
>> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org
>> >>>> ] On Behalf Of Russell.Mosemann at cune.org
>> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
>> >>>> To: freeswitch-users at lists.freeswitch.org
>> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>> >>>>
>> >>>> Chris Fowler <chris at fowler.cc> said:
>> >>>>
>> >>>>> There were no special tricks; you do need to modify/override the
>> >>>>> following with the box's Elastic IP (EIP).
>> >>>>
>> >>>> This would be helpful to have in the wiki along with any other tips
>> >>>> for
>> >>>> your virtual environment. :-)
>> >>>>
>> >>>> --
>> >>>> Russell Mosemann
>> >>>>
>> >>>>
>> >>>>
>> >>>> ________________________________________________________
>> >>>> Concordia University, Nebraska
>> >>>> See http://www.cune.edu/ for the latest news and events!
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Sincerely,
>> >>>
>> >>> Giovanni Maruzzelli
>> >>> Cell : +39-347-2665618
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > Cell : +39-347-2665618
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:14:36 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:14:36 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] Call transfer using
	eventsockets with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
References: <Acq8U5J/iQUC0nAxSbWQSsxFzefS/Q==>
	<3181A30B8C35AB4AA8577B78DDF46138068F6214@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072014h4368cb2x1a02c0fb5953504d@mail.gmail.com>

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

Regards,
Jo?o Mesquita


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

>  Hi,
>
>
>
> Is there a way to transfer call with FSComm. I tried different ways using
> eventsockets but failed.
>
>
>
> Tried the scenario as follows:
>
> Received a call from 1001 to FSComm (registered with 1002) and is answered
> (Unique-id is 784dd690-6d0a-47de-b5f4-b923264581a5). Made a call from FSComm
> to extension 1003 and is answered (Unique-id is
> ec7a3bdd-c265-41fb-8cef-b897d2e8bf62). Now I want to transfer this call to
> 1001 and FSComm should be out of loop.
>
>
>
> a.       Tried with command ?api uuid_transfer
> 784dd690-6d0a-47de-b5f4-b923264581a5 ?both park inline? but failed. With
> this call at 1001 is dropped and call between FSComm and 1003 is still
> there.
>
> b.      Tried with command ?api uuid_bridge
> 784dd690-6d0a-47de-b5f4-b923264581a5 ec7a3bdd-c265-41fb-8cef-b897d2e8bf62?.
> With this all calls are dropped.
>
>
>
> Could someone let me know if there is any procedure for call transfer
> (using event sockets).
>
>
>
> Thanks & Regards
>
> Suneel
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:15:19 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:15:19 -0300
Subject: [Freeswitch-users] FScomm
In-Reply-To: <13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705>
	<COL117-W5563292C886257AFC56548B9420@phx.gbl>
	<D64783281177440B9094B8181E55EFA7@MOBILEE1705>
	<COL117-W429F3BED9F781A58264238B9420@phx.gbl>
	<11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705>
	<BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com>
	<D262917F3F4943D49370D0CC015A7582@MOBILEE1705>
	<ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com>
	<13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
Message-ID: <ebbc2d551003072015t7537eae4g6564c1286fb1fd3b@mail.gmail.com>

If I can access your system over SSH I can probably figure this out. Is this
possible?

Jo?o Mesquita


On Wed, Mar 3, 2010 at 3:26 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi Joao,
>
> thanks for your answer.
> I have a standard RPM QT last update for Fedora10 64bits.
> so I really don't know what path FScomm needs
>
> Regards
>
> Franck
>
> ----- Original Message -----
> *From:* Jo?o Mesquita <jmesquita at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 03, 2010 1:00 AM
> *Subject:* Re: [Freeswitch-users] FScomm
>
> This is definitely a problem with your Qt installation. You need to be able
> to find the libs to link it, otherwise, unresolved symbols everywhere!
>
> Regards,
> Jo?o Mesquita
> FSComm Developer
>
>
> On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:
>
>>
>> ----- Original Message -----
>> From: "Michael Jerris" <mike at jerris.com>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, February 24, 2010 3:40 AM
>>  Subject: Re: [Freeswitch-users] FScomm
>>
>>
>>
>> On Feb 23, 2010, at 6:18 PM, Madovsky wrote:
>>
>> >
>> > ----- Original Message -----
>> > From: Jeff Lenk
>> > To: freeswitch-users at lists.freeswitch.org
>> > Sent: Tuesday, February 23, 2010 5:05 PM
>> > Subject: Re: [Freeswitch-users] FScomm
>> >
>> >
>> > http://wiki.freeswitch.org/wiki/FSComm#Linux
>> >
>> > you must run those from the FSComm directory
>> >
>>
>> ?.
>>
>> >
>> > It's what I did,
>> > but from FS trunk, inside fscomm directory,
>> > there s only
>> >
>> > account.cpp  conf                fshost.h        mainwindow.ui
>> > resources.qrc
>> > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
>> > call.cpp     FSComm.pro          mainwindow.cpp  preferences
>> > call.h       fshost.cpp          mainwindow.h    resources
>> >
>>
>>
>> Read those installation instructions again and do them step by step, you
>> skipped one.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> Ok now I have
>>
>> [root at node250 fscomm]# qmake
>> WARNING: Found potential symbol conflict of mainwindow.cpp
>> (mainwindow.cpp)
>> in SOURCES
>> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
>> HEADERS
>> WARNING: Found potential symbol conflict of prefdialog.cpp
>> (preferences/prefdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of prefdialog.h
>> (preferences/prefdialog.h) in HEADERS
>> WARNING: Found potential symbol conflict of accountdialog.cpp
>> (preferences/accountdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of accountdialog.h
>> (preferences/accountdialog.h) in HEADERS
>>
>> [root at node250 fscomm]# qmake
>> WARNING: Found potential symbol conflict of mainwindow.cpp
>> (mainwindow.cpp)
>> in SOURCES
>> WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
>> HEADERS
>> WARNING: Found potential symbol conflict of prefdialog.cpp
>> (preferences/prefdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of prefdialog.h
>> (preferences/prefdialog.h) in HEADERS
>> WARNING: Found potential symbol conflict of accountdialog.cpp
>> (preferences/accountdialog.cpp) in SOURCES
>> WARNING: Found potential symbol conflict of accountdialog.h
>> (preferences/accountdialog.h) in HEADERS
>> [root at node250 fscomm]# make
>> Makefile:278: warning: overriding commands for target `prefdialog.o'
>> Makefile:215: warning: ignoring old commands for target `prefdialog.o'
>> Makefile:285: warning: overriding commands for target `accountdialog.o'
>> Makefile:234: warning: ignoring old commands for target `accountdialog.o'
>> Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
>> Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
>> Makefile:323: warning: overriding commands for target
>> `moc_accountdialog.o'
>> Makefile:307: warning: ignoring old commands for target
>> `moc_accountdialog.o'
>> Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
>> Makefile:326: warning: ignoring old commands for target
>> `moc_mainwindow.cpp'
>> Makefile:350: warning: overriding commands for target
>> `preferences/moc_prefdialog.cpp'
>> Makefile:332: warning: ignoring old commands for target
>> `preferences/moc_prefdialog.cpp'
>> Makefile:353: warning: overriding commands for target
>> `preferences/moc_accountdialog.cpp'
>> Makefile:341: warning: ignoring old commands for target
>> `preferences/moc_accountdialog.cpp'
>> g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2
>> -fexceptions
>>  -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic
>>  -DQT_NO_DEBUG
>>  -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT
>> -I/usr/lib64/qt-3.3/mkspecs/default
>>  -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src
>> -I/usr/lib64/qt-3.3/include
>>  -o main.o main.cpp
>> main.cpp:31:25: error: QSplashScreen: No such file or directory
>> In file included from main.cpp:32:
>> mainwindow.h:34:23: error: QMainWindow: No such file or directory
>> mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
>> mainwindow.h:36:25: error: QSignalMapper: No such file or directory
>> mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
>> In file included from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./fshost.h:32:19: error: QThread: No such file or directory
>> ./fshost.h:33:17: error: QHash: No such file or directory
>> ./fshost.h:34:26: error: QSharedPointer: No such file or directory
>> In file included from ./fshost.h:36,
>>                 from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./call.h:32:18: error: QtCore: No such file or directory
>> ./call.h:33:19: error: QString: No such file or directory
>> In file included from mainwindow.h:42,
>>                 from main.cpp:32:
>> preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
>> preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
>> directory
>> preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
>> In file included from ./fshost.h:37,
>>                 from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./account.h:18: error: expected constructor, destructor, or type
>> conversion
>> before ?static?
>> In file included from mainwindow.h:39,
>>                 from main.cpp:32:
>> ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
>> /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
>> QThread?
>> ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:46: error: expected ?;? before ?<? token
>> ./fshost.h:47: error: expected `;' before ?QSharedPointer?
>> ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:47: error: expected ?;? before ?<? token
>> ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:48: error: template argument 1 is invalid
>> ./fshost.h:48: error: expected unqualified-id before ?>? token
>> ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:49: error: expected ?;? before ?<? token
>> ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:50: error: expected ?;? before ?<? token
>> ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
>> no type
>> ./fshost.h:51: error: expected ?;? before ?<? token
>> ./fshost.h:52: error: ?QSharedPointer? has not been declared
>> ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:60: error: ?QSharedPointer? has not been declared
>> ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:61: error: ?QSharedPointer? has not been declared
>> ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:62: error: ?QSharedPointer? has not been declared
>> ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:63: error: ?QSharedPointer? has not been declared
>> ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:64: error: ?QSharedPointer? has not been declared
>> ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:65: error: ?QSharedPointer? has not been declared
>> ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:66: error: ?QSharedPointer? has not been declared
>> ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:67: error: ?QSharedPointer? has not been declared
>> ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:71: error: ?QSharedPointer? has not been declared
>> ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
>> ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:78: error: template argument 2 is invalid
>> ./fshost.h:78: error: expected unqualified-id before ?>? token
>> ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
>> ./fshost.h:79: error: template argument 2 is invalid
>> ./fshost.h:79: error: expected unqualified-id before ?>? token
>> ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
>> In file included from mainwindow.h:42,
>>                 from main.cpp:32:
>> preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
>> QDialog?
>> /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
>> QDialog?
>> preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
>> ?QSettings? with no type
>> preferences/prefdialog.h:31: error: expected ?;? before ?*? token
>> In file included from main.cpp:32:
>> mainwindow.h:48: error: expected class-name before ?{? token
>> mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
>> mainwindow.h:71: error: ?QSharedPointer? has not been declared
>> mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:72: error: ?QSharedPointer? has not been declared
>> mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:73: error: ?QSharedPointer? has not been declared
>> mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:74: error: ?QSharedPointer? has not been declared
>> mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:75: error: ?QSharedPointer? has not been declared
>> mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:78: error: ?QSharedPointer? has not been declared
>> mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:79: error: ?QSharedPointer? has not been declared
>> mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:80: error: ?QSharedPointer? has not been declared
>> mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
>> mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
>> mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
>> mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper?
>> with
>> no type
>> mainwindow.h:86: error: expected ?;? before ?*? token
>> mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
>> with no type
>> mainwindow.h:88: error: expected ?;? before ?*? token
>> main.cpp: In function ?int main(int, char**)?:
>> main.cpp:41: error: variable ?QPixmap image? has initializer but
>> incomplete
>> type
>> main.cpp:42: error: ?QSplashScreen? was not declared in this scope
>> main.cpp:42: error: ?splash? was not declared in this scope
>> main.cpp:42: error: expected type-specifier before ?QSplashScreen?
>> main.cpp:42: error: expected `;' before ?QSplashScreen?
>> main.cpp:48: error: no matching function for call to
>> ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
>> /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
>> QObject::connect(const QObject*, const char*, const QObject*, const char*,
>> Qt::ConnectionType)
>> /usr/include/QtCore/qobject.h:307: note:                 bool
>> QObject::connect(const QObject*, const char*, const char*,
>> Qt::ConnectionType) const
>> main.cpp:49: error: ?class FSHost? has no member named ?start?
>> main.cpp:41: warning: unused variable ?image?
>> ./fshost.h: At global scope:
>> ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)?
>> defined
>> but not used
>> make: *** [main.o] Error 1
>>
>>
>> Any idea ?
>>
>> Thx
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar  7 20:18:59 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 8 Mar 2010 01:18:59 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] FSComm basic issue
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
References: <Acq3BRPNPSeAcwujTl+n9yQujbXGcA==>
	<3181A30B8C35AB4AA8577B78DDF4613806886903@nickel.mettonigroup.com>
Message-ID: <ebbc2d551003072018r294ee37ag4774121fea30e689@mail.gmail.com>

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

Inline...

JM

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

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

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

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

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

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


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


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

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

>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *************************************************************************
> Please consider the environment before printing this e-mail
> *************************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.  http://www.mettoni.com
>
> Mettoni Ltd
> Registered in England and Wales: 4485956
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From freeswitch.org at todandlorna.com  Sun Mar  7 21:28:46 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Sun, 07 Mar 2010 22:28:46 -0700
Subject: [Freeswitch-users] Conference Recording
In-Reply-To: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
References: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
Message-ID: <4B948B0E.8050909@todandlorna.com>

For the whole thing?  I know parts were recorded by different people.  
I'm not sure about the whole thing.

-Tod Hansmann

On 3/7/2010 5:28 PM, Josh Rivers wrote:
> On last week's conference call, someone was recording, did that 
> recording get put online somewhere?
>
> Josh
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From infos at madovsky.org  Sun Mar  7 22:20:01 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 01:20:01 -0500
Subject: [Freeswitch-users] registration in a DB shared mode
Message-ID: <AF56DAE1642E48AB848D8F5761DAE125@MOBILEE1705>

I set 2 freeswitch on 2 nodes with a shared DB,
set all configs for shared db with ODBC postgresql,
but if I call an extension registered from the another node 
voicemail answer. I missed to set something somewhere,
if anyone can help it would be great

Regards

Franck
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From jingwei.yang at gmail.com  Mon Mar  8 00:43:01 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 8 Mar 2010 16:43:01 +0800
Subject: [Freeswitch-users] How to set background music for application park
Message-ID: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>

Hello,

I'm using this command to put each party of a call to hold:

uuid_transfer <uuid> -both 'park' inline

It works perfectly except none of them can hear the background music. May I
know how to set the music in this case? And is it possible to set different
music for different parties?

Thanks,
-Jingwei
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From gavin.henry at gmail.com  Mon Mar  8 01:34:34 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 09:34:34 +0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
Message-ID: <13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>

We use a 256mb xen vm for customers hosted pbx platforms for up to 20
users. Everything is fine re conferencing and voicemails etc. We also
use the -vm flag and this is driven by FusionPBX

Each customer gets their own vm so not much load.

Thanks,

Gavin.

On 08/03/2010, Rob Forman <rob4manhere at gmail.com> wrote:
> Nice to see so many people using it in so many different ways.  For those
> using ec2 or xen, have you noticed this in your logs?
>
> 2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating timer,
> please wait...
> 2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer resolution of
> 4000 microseconds detected!
> Do you have your kernel timer set to higher than 1 kHz? You may experience
> audio problems.
>
>
> "You may experience audio problems."  Seems pretty clear.  Can that just be
> ignored?
>
> Rob
>
>
> On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
>> wrote:
>
>> I used to do tests on conferencing using linode.com, which is Xen VMs
>> without issues, although there were not much load on it.
>>
>> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit), seems
>>> pretty well. Of course no much load.
>>>
>>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>>> > On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman <rob4manhere at gmail.com>
>>> wrote:
>>> >>  >  Cheers, Chris.
>>> >>  >  --
>>> >>  >  RightScale, Inc.
>>> >>
>>> >> I doubt they'll be switching any time soon.  I think his employer IS
>>> >> RightScale :)
>>> >
>>> > Oooops, I'll learn to pay attention to signatures :)
>>> >
>>> >
>>> >>
>>> >>
>>> >> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>> >>
>>> >>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc> wrote:
>>> >>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>>> >>>> have an objection if I create a "How to use FreeSWITCH on EC2 with
>>> >>>> RightScale's management dashboard"?
>>> >>>>
>>> >>>> There are some significant differences to the configuration when
>>> >>>> using RightScale vs. bare bones AWS; and given this is on my
>>> >>>> employer's time I need to keep it relevant...
>>> >>>
>>> >>> Chris,
>>> >>> I know can be more work, but the best would be to let know the
>>> >>> differences too, and how it would be using bare bone AWS. That's also
>>> >>> good for your employer, if he decide to switch back from
>>> >>> RightScale...
>>> >>> :)
>>> >>>
>>> >>> -giovanni
>>> >>>
>>> >>>
>>> >>>>
>>> >>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10"; this is
>>> >>>> a public AMI (you don't need to be a RightScale customer to use it)
>>> >>>>
>>> >>>> Cheers, Chris.
>>> >>>>
>>> >>>> -----Original Message-----
>>> >>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org
>>> >>>> ] On Behalf Of Russell.Mosemann at cune.org
>>> >>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>> >>>> To: freeswitch-users at lists.freeswitch.org
>>> >>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>> >>>>
>>> >>>> Chris Fowler <chris at fowler.cc> said:
>>> >>>>
>>> >>>>> There were no special tricks; you do need to modify/override the
>>> >>>>> following with the box's Elastic IP (EIP).
>>> >>>>
>>> >>>> This would be helpful to have in the wiki along with any other tips
>>> >>>> for
>>> >>>> your virtual environment. :-)
>>> >>>>
>>> >>>> --
>>> >>>> Russell Mosemann
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> ________________________________________________________
>>> >>>> Concordia University, Nebraska
>>> >>>> See http://www.cune.edu/ for the latest news and events!
>>> >>>>
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> --
>>> >>> Sincerely,
>>> >>>
>>> >>> Giovanni Maruzzelli
>>> >>> Cell : +39-347-2665618
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Sincerely,
>>> >
>>> > Giovanni Maruzzelli
>>> > Cell : +39-347-2665618
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From jaybinks at gmail.com  Mon Mar  8 03:30:48 2010
From: jaybinks at gmail.com (Jay Binks)
Date: Mon, 8 Mar 2010 21:30:48 +1000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<20100303181909.DEFD22E73C8@cuneorg-email.cune.pri>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
	<13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
Message-ID: <70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>

The qn here is do you bypass media or do you do complete topology  
hiding ??

Sip only is kind of a given ,  it gets harder if the virtualized pbx  
does media.

J



On 08/03/2010, at 19:34, Gavin Henry <gavin.henry at gmail.com> wrote:

> We use a 256mb xen vm for customers hosted pbx platforms for up to 20
> users. Everything is fine re conferencing and voicemails etc. We also
> use the -vm flag and this is driven by FusionPBX
>
> Each customer gets their own vm so not much load.
>
> Thanks,
>
> Gavin.
>
> On 08/03/2010, Rob Forman <rob4manhere at gmail.com> wrote:
>> Nice to see so many people using it in so many different ways.  For  
>> those
>> using ec2 or xen, have you noticed this in your logs?
>>
>> 2010-03-08 02:29:50.472745 [CONSOLE] switch_time.c:1021 Calibrating  
>> timer,
>> please wait...
>> 2010-03-08 02:29:50.472851 [WARNING] switch_time.c:206 Timer  
>> resolution of
>> 4000 microseconds detected!
>> Do you have your kernel timer set to higher than 1 kHz? You may  
>> experience
>> audio problems.
>>
>>
>> "You may experience audio problems."  Seems pretty clear.  Can that  
>> just be
>> ignored?
>>
>> Rob
>>
>>
>> On Thu, Mar 4, 2010 at 8:55 AM, Fernando Testa <testa at voicetechnology.com.br
>>> wrote:
>>
>>> I used to do tests on conferencing using linode.com, which is Xen  
>>> VMs
>>> without issues, although there were not much load on it.
>>>
>>> On Wed, Mar 3, 2010 at 10:04 PM, Seven Du <dujinfang at gmail.com>  
>>> wrote:
>>>
>>>> we run our testing environment on Xen VMs (ubuntu 8.04 32bit),  
>>>> seems
>>>> pretty well. Of course no much load.
>>>>
>>>> 2010/3/4 Giovanni Maruzzelli <gmaruzz at celliax.org>:
>>>>> On Wed, Mar 3, 2010 at 10:09 PM, Rob Forman  
>>>>> <rob4manhere at gmail.com>
>>>> wrote:
>>>>>>> Cheers, Chris.
>>>>>>> --
>>>>>>> RightScale, Inc.
>>>>>>
>>>>>> I doubt they'll be switching any time soon.  I think his  
>>>>>> employer IS
>>>>>> RightScale :)
>>>>>
>>>>> Oooops, I'll learn to pay attention to signatures :)
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>> On Mar 3, 2010, at 3:00 PM, Giovanni Maruzzelli wrote:
>>>>>>
>>>>>>> On Wed, Mar 3, 2010 at 9:48 PM, Chris Fowler <chris at fowler.cc>  
>>>>>>> wrote:
>>>>>>>> Re: Wiki - I'd love to give back to the community;  Does anyone
>>>>>>>> have an objection if I create a "How to use FreeSWITCH on EC2  
>>>>>>>> with
>>>>>>>> RightScale's management dashboard"?
>>>>>>>>
>>>>>>>> There are some significant differences to the configuration  
>>>>>>>> when
>>>>>>>> using RightScale vs. bare bones AWS; and given this is on my
>>>>>>>> employer's time I need to keep it relevant...
>>>>>>>
>>>>>>> Chris,
>>>>>>> I know can be more work, but the best would be to let know the
>>>>>>> differences too, and how it would be using bare bone AWS.  
>>>>>>> That's also
>>>>>>> good for your employer, if he decide to switch back from
>>>>>>> RightScale...
>>>>>>> :)
>>>>>>>
>>>>>>> -giovanni
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> The base AMI used is "RightImage CentOS_5.4_i386_v4.4.10";  
>>>>>>>> this is
>>>>>>>> a public AMI (you don't need to be a RightScale customer to  
>>>>>>>> use it)
>>>>>>>>
>>>>>>>> Cheers, Chris.
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org
>>>>>>>> ] On Behalf Of Russell.Mosemann at cune.org
>>>>>>>> Sent: Wednesday, March 03, 2010 10:19 AM
>>>>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>>>>> Subject: Re: [Freeswitch-users] Virtualized FreeSWITCH
>>>>>>>>
>>>>>>>> Chris Fowler <chris at fowler.cc> said:
>>>>>>>>
>>>>>>>>> There were no special tricks; you do need to modify/override  
>>>>>>>>> the
>>>>>>>>> following with the box's Elastic IP (EIP).
>>>>>>>>
>>>>>>>> This would be helpful to have in the wiki along with any  
>>>>>>>> other tips
>>>>>>>> for
>>>>>>>> your virtual environment. :-)
>>>>>>>>
>>>>>>>> --
>>>>>>>> Russell Mosemann
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> ________________________________________________________
>>>>>>>> Concordia University, Nebraska
>>>>>>>> See http://www.cune.edu/ for the latest news and events!
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> -- 
> Sent from my mobile device
>
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From jbrucehopkins at gmail.com  Mon Mar  8 04:56:12 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 12:56:12 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
Message-ID: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>

Hi,

I wonder if some kind soul might be able please to help me by pointing out
what I am doing wrong with my doublenat configuration.

I have incoming calls from WAN --> LAN working fine, but when I try to call
from LAN -->WAN, the Freeswitch CLI says:


"[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type
[error] cause: [USER_NOT_REGISTERED]"


The setup is essentially:


phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT
(public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)


Phone_1 is able to register with Freeswitch using a "doublenat" sip profile
I set up using port 5090, created from the default external profile with the
port changed. phone_1 can initiate calls to phone_2 fine with no problems
with audio - everything works just fine in that direction.

However when I try to call from phone_2 --> phone_1 I get the message "The
person at extension 1002 is not available.  Record your message at the
tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not
registered.

In trying to solve this I have run a Wireshark trace onthe PC running the
phone_1 softphone to check that the SIP registration seems to go as I
understand it should.  the dialog is as follows:

phone_1 --> FS             request: REGISTER
FS         -->  phone_1    status: 401 Unauthorised
phone_1 --> FS             request: REGISTER
FS         --> phone_1     status: 200 OK
phone_1 --> FS             request: SUBSCRIBE
FS         --> phone_1     status: 405 Method Not Allowed


There does appear to be a difference here to what happens when phone_2 (on
the same LAN as Freeswitch) registers.  With phone_2 I see:

phone_2 --> FS             request: REGISTER
FS         --> phone_2     status: 401 Unauthorised
phone_2 --> FS             request: REGISTER
FS         --> phone_2     status  200 OK
phone_2 --> FS             request: SUBSCRIBE
FS         --> phone_2     status: 202 Accepted
FS         --> phone_2     request: NOTIFY
phone_2 --> FS             status: 200 OK



More configuration details:

In conf/sip_profiles/doublenat.xml I have not uncommented <param
name="force-register-domain" value="$$(domain)"/>

In conf/sip_profiles/doublenat.xml I have tried setting the context to each
of public and default.  This does not seem to make a difference.

In conf/directory/doublenat.xml I have set the domain name to be the domain
name registerd for Fs's external IP with DNS, and used by phone_1.

In conf/directory/doublenat.xml I have tried both of the following 9I  don't
know if this is relevant)

1. Firstly with phone_1 registering to an extension in the default
configuration, held in conf/directory/default/:

<groups>
    <group name="default">
       <users>
          <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
       </users>
    </group>

2.  Then I tried moving the .xml file for phone_1's extension to
conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:

<groups>
    <group name="doublenat">
       <users>
          <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
       </users>
     </group>

Unfortunately this does not make a difference to the issue though.

If anybody had the time to tell me what I am doing wrong I would be hugely
grateful !

Many thanks in advance
Bruce
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From jbrucehopkins at gmail.com  Mon Mar  8 05:25:35 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 13:25:35 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
Message-ID: <d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>

Hi Anthony,

Many, many apologies to take so long to reply - I failed to notice your
response.  Sorry !

It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
then does not give a warning.

Best wishes
Bruce

On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> we are still wary about 5.3 due to bugs reported in libc.
> What number does it say it detected for the gap?
>
>
> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>
>> OK - I've realised I do get the same warning with CentOS 5.3, it just goes
>> past more quickly so I didn't see it.  Maybe it is just the hardware ....
>>
>>
>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I wonder if anyone would be able to advise please:
>>>
>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
>>> start FreeSWITCH that
>>>
>>> "Abnormally large timer gap detected"
>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>> experience audio problems".
>>>
>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>> to Centos5.4 was a yum update on the 5.3 build.
>>>
>>> I guess the warning comes from here:
>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>
>>> This is all on pretty low spec hardware - a couple of different Dell
>>> optiplex p4's I use for testing.
>>>
>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>> use such crummy hardware?!
>>>
>>> Many thanks in advance
>>> Bruce
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From shaheryarkh at googlemail.com  Mon Mar  8 05:38:42 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 18:38:42 +0500
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
Message-ID: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>

Hi,

I am trying to enable ZRTP support in FS, following the instructions given
at http://wiki.freeswitch.org/wiki/ZRTP.

Upon executing Step 4, i get following error, afterwords it starts some
configure script.

root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
tar: Error is not recoverable: exiting now
tar: Child returned status 2
tar: Error exit delayed from previous errors
cd: 3: can't cd to libzrtp-0.81.514
build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such
file
cd: 5: can't cd to projects/gnu/
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for a thread-safe mkdir -p... /bin/mkdir -p
checking for gawk... no
checking for mawk... mawk
checking whether make sets $(MAKE)... yes
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
....

I have no idea, how to fix this. Please help.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Mon Mar  8 06:26:32 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:26:32 -0600
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
In-Reply-To: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
References: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
Message-ID: <920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>

step 1. get the latest libzrtp from their website.  Step 2 you don't need the patches now. Step 3. compile and install the lib with --enable-pic

/b


On Mar 8, 2010, at 7:38 AM, Muhammad Shahzad wrote:

> Hi,
> 
> I am trying to enable ZRTP support in FS, following the instructions given at http://wiki.freeswitch.org/wiki/ZRTP.
> 
> Upon executing Step 4, i get following error, afterwords it starts some configure script.
> 
> root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
> tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
> tar: Error is not recoverable: exiting now
> tar: Child returned status 2
> tar: Error exit delayed from previous errors
> cd: 3: can't cd to libzrtp-0.81.514
> build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such file
> cd: 5: can't cd to projects/gnu/
> checking for a BSD-compatible install... /usr/bin/install -c
> checking whether build environment is sane... yes
> checking for a thread-safe mkdir -p... /bin/mkdir -p
> checking for gawk... no
> checking for mawk... mawk
> checking whether make sets $(MAKE)... yes
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> ....
> 
> I have no idea, how to fix this. Please help.
> 
> Thank you.
> 
> 
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Mar  8 06:28:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:28:05 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
Message-ID: <4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>

Step 1. Revert everything back to SVN trunk.
Step 2. Enable UPNP/NATPMP on both routers.
Step 3. It should just work.  The doublenat profile is NO LONGER NEEDED.

If you can't enable UPNP/NATPMP then by all means static map the ports correctly or put the FreeSWITCH box in the DMZ on both ends.  It should just work.

/b

On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:

> Hi,
>  
> I wonder if some kind soul might be able please to help me by pointing out what I am doing wrong with my doublenat configuration.
>  
> I have incoming calls from WAN --> LAN working fine, but when I try to call from LAN -->WAN, the Freeswitch CLI says:
>  
>  
> "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]"
>  
>  
> The setup is essentially:
>  
>  
> phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
>  
>  
> Phone_1 is able to register with Freeswitch using a "doublenat" sip profile I set up using port 5090, created from the default external profile with the port changed. phone_1 can initiate calls to phone_2 fine with no problems with audio - everything works just fine in that direction. 
>  
> However when I try to call from phone_2 --> phone_1 I get the message "The person at extension 1002 is not available.  Record your message at the tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not registered.
>  
> In trying to solve this I have run a Wireshark trace onthe PC running the phone_1 softphone to check that the SIP registration seems to go as I understand it should.  the dialog is as follows:
>  
> phone_1 --> FS             request: REGISTER
> FS         -->  phone_1    status: 401 Unauthorised
> phone_1 --> FS             request: REGISTER
> FS         --> phone_1     status: 200 OK
> phone_1 --> FS             request: SUBSCRIBE
> FS         --> phone_1     status: 405 Method Not Allowed
>  
>  
> There does appear to be a difference here to what happens when phone_2 (on the same LAN as Freeswitch) registers.  With phone_2 I see:
>  
> phone_2 --> FS             request: REGISTER
> FS         --> phone_2     status: 401 Unauthorised
> phone_2 --> FS             request: REGISTER
> FS         --> phone_2     status  200 OK
> phone_2 --> FS             request: SUBSCRIBE
> FS         --> phone_2     status: 202 Accepted
> FS         --> phone_2     request: NOTIFY
> phone_2 --> FS             status: 200 OK
>  
>  
>  
> More configuration details:
>  
> In conf/sip_profiles/doublenat.xml I have not uncommented <param name="force-register-domain" value="$$(domain)"/>
>  
> In conf/sip_profiles/doublenat.xml I have tried setting the context to each of public and default.  This does not seem to make a difference.
>  
> In conf/directory/doublenat.xml I have set the domain name to be the domain name registerd for Fs's external IP with DNS, and used by phone_1.
>  
> In conf/directory/doublenat.xml I have tried both of the following 9I  don't know if this is relevant)
>  
> 1. Firstly with phone_1 registering to an extension in the default configuration, held in conf/directory/default/:
>  
> <groups>
>     <group name="default">
>        <users>
>           <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
>        </users>
>     </group>
>  
> 2.  Then I tried moving the .xml file for phone_1's extension to conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
>  
> <groups>
>     <group name="doublenat">
>        <users>
>           <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
>        </users>
>      </group>
>  
> Unfortunately this does not make a difference to the issue though.
>  
> If anybody had the time to tell me what I am doing wrong I would be hugely grateful !
>  
> Many thanks in advance
> Bruce
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Mar  8 06:30:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 08:30:09 -0600
Subject: [Freeswitch-users] How to set background music for application
	park
In-Reply-To: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>
References: <13529f9d1003080043u34f870c8vf7786ec8296f6be5@mail.gmail.com>
Message-ID: <75652D86-1C32-4FF7-B9AE-B328EC069BAB@freeswitch.org>

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park

/b

On Mar 8, 2010, at 2:43 AM, Jingwei Yang wrote:

> Hello,
> 
> I'm using this command to put each party of a call to hold:
> 
> uuid_transfer <uuid> -both 'park' inline
> 
> It works perfectly except none of them can hear the background music. May I know how to set the music in this case? And is it possible to set different music for different parties?
> 
> Thanks,
> -Jingwei
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




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

Hello FreeSWITCHers,

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

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

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

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

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From shaheryarkh at googlemail.com  Mon Mar  8 06:53:55 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 19:53:55 +0500
Subject: [Freeswitch-users] FS ZRTP Support: Build Error
In-Reply-To: <920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>
References: <b16156851003080538p7079d0fcsfa47e862d14be9a1@mail.gmail.com>
	<920C4D73-C767-43CB-9A5C-1F88E4E6CC25@freeswitch.org>
Message-ID: <b16156851003080653w7db644e8q7cc7277b42a2bf4d@mail.gmail.com>

The file i downloaded from their site is named as zfone-linux.tar. Let me
install that and rebuild FS to see if zrtp support compiles.

Thank you.


On Mon, Mar 8, 2010 at 7:26 PM, Brian West <brian at freeswitch.org> wrote:

> step 1. get the latest libzrtp from their website.  Step 2 you don't need
> the patches now. Step 3. compile and install the lib with --enable-pic
>
> /b
>
>
> On Mar 8, 2010, at 7:38 AM, Muhammad Shahzad wrote:
>
> Hi,
>
> I am trying to enable ZRTP support in FS, following the instructions given
> at http://wiki.freeswitch.org/wiki/ZRTP.
>
> Upon executing Step 4, i get following error, afterwords it starts some
> configure script.
>
> root at slave:/usr/src/svn-src/freeswitch# sh build/buildzrtp.sh
> tar: libzrtp-0.81.514.tar.gz: Cannot open: No such file or directory
> tar: Error is not recoverable: exiting now
> tar: Child returned status 2
> tar: Error exit delayed from previous errors
> cd: 3: can't cd to libzrtp-0.81.514
> build/buildzrtp.sh: 4: cannot open ../patches/zrtp_bnlib_pic.diff: No such
> file
> cd: 5: can't cd to projects/gnu/
> checking for a BSD-compatible install... /usr/bin/install -c
> checking whether build environment is sane... yes
> checking for a thread-safe mkdir -p... /bin/mkdir -p
> checking for gawk... no
> checking for mawk... mawk
> checking whether make sets $(MAKE)... yes
> checking build system type... i686-pc-linux-gnu
> checking host system type... i686-pc-linux-gnu
> checking for gcc... gcc
> ....
>
> I have no idea, how to fix this. Please help.
>
> Thank you.
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From jbrucehopkins at gmail.com  Mon Mar  8 07:03:55 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 15:03:55 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
Message-ID: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>

Many thanks for getting back to me with help so quickly Brian.

I was using 1.0.5-20100302-0400 so will update to trunk and do as you
suggest.

thanks again
Bruce

On 8 March 2010 14:28, Brian West <brian at freeswitch.org> wrote:

> Step 1. Revert everything back to SVN trunk.
> Step 2. Enable UPNP/NATPMP on both routers.
> Step 3. It should just work.  The doublenat profile is NO LONGER NEEDED.
>
> If you can't enable UPNP/NATPMP then by all means static map the ports
> correctly or put the FreeSWITCH box in the DMZ on both ends.  It should just
> work.
>
> /b
>
> On Mar 8, 2010, at 6:56 AM, Bruce Hopkins wrote:
>
> > Hi,
> >
> > I wonder if some kind soul might be able please to help me by pointing
> out what I am doing wrong with my doublenat configuration.
> >
> > I have incoming calls from WAN --> LAN working fine, but when I try to
> call from LAN -->WAN, the Freeswitch CLI says:
> >
> >
> > "[ERR} switch_ivr_originate.c.2389 CAnnot create outgoing channel of type
> [error] cause: [USER_NOT_REGISTERED]"
> >
> >
> > The setup is essentially:
> >
> >
> > phone_1 (192.168.a.b) --> NAT (public ip 1.2.3.4) --> Internet --> NAT
> (public ip 5.6.7.8) --> Freeswitch (192.168.x.y) --> phone_2 (192.168.x.z)
> >
> >
> > Phone_1 is able to register with Freeswitch using a "doublenat" sip
> profile I set up using port 5090, created from the default external profile
> with the port changed. phone_1 can initiate calls to phone_2 fine with no
> problems with audio - everything works just fine in that direction.
> >
> > However when I try to call from phone_2 --> phone_1 I get the message
> "The person at extension 1002 is not available.  Record your message at the
> tone."  Meanwhile the CLI shows that Freeswitch says phone_1 is not
> registered.
> >
> > In trying to solve this I have run a Wireshark trace onthe PC running the
> phone_1 softphone to check that the SIP registration seems to go as I
> understand it should.  the dialog is as follows:
> >
> > phone_1 --> FS             request: REGISTER
> > FS         -->  phone_1    status: 401 Unauthorised
> > phone_1 --> FS             request: REGISTER
> > FS         --> phone_1     status: 200 OK
> > phone_1 --> FS             request: SUBSCRIBE
> > FS         --> phone_1     status: 405 Method Not Allowed
> >
> >
> > There does appear to be a difference here to what happens when phone_2
> (on the same LAN as Freeswitch) registers.  With phone_2 I see:
> >
> > phone_2 --> FS             request: REGISTER
> > FS         --> phone_2     status: 401 Unauthorised
> > phone_2 --> FS             request: REGISTER
> > FS         --> phone_2     status  200 OK
> > phone_2 --> FS             request: SUBSCRIBE
> > FS         --> phone_2     status: 202 Accepted
> > FS         --> phone_2     request: NOTIFY
> > phone_2 --> FS             status: 200 OK
> >
> >
> >
> > More configuration details:
> >
> > In conf/sip_profiles/doublenat.xml I have not uncommented <param
> name="force-register-domain" value="$$(domain)"/>
> >
> > In conf/sip_profiles/doublenat.xml I have tried setting the context to
> each of public and default.  This does not seem to make a difference.
> >
> > In conf/directory/doublenat.xml I have set the domain name to be the
> domain name registerd for Fs's external IP with DNS, and used by phone_1.
> >
> > In conf/directory/doublenat.xml I have tried both of the following 9I
>  don't know if this is relevant)
> >
> > 1. Firstly with phone_1 registering to an extension in the default
> configuration, held in conf/directory/default/:
> >
> > <groups>
> >     <group name="default">
> >        <users>
> >           <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
> >        </users>
> >     </group>
> >
> > 2.  Then I tried moving the .xml file for phone_1's extension to
> conf/drectory/doublenat/ and changed conf/directory/doublenat.xml to show:
> >
> > <groups>
> >     <group name="doublenat">
> >        <users>
> >           <X-PRE-PROCESS cmd="include" data="doublenat/*.xml"/>
> >        </users>
> >      </group>
> >
> > Unfortunately this does not make a difference to the issue though.
> >
> > If anybody had the time to tell me what I am doing wrong I would be
> hugely grateful !
> >
> > Many thanks in advance
> > Bruce
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From srinivas.ksvreddy at gmail.com  Mon Mar  8 07:08:34 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 8 Mar 2010 20:38:34 +0530
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
Message-ID: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>

Hi,

can any body tried to connect cisco router from Freeswitch, please help me.

-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Mar  8 07:15:01 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:15:01 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
Message-ID: <C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>

Well sounds like you need to sync in your configs too.

/b

On Mar 8, 2010, at 9:03 AM, Bruce Hopkins wrote:

> Many thanks for getting back to me with help so quickly Brian.
> 
> I was using 1.0.5-20100302-0400 so will update to trunk and do as you suggest.
> 
> thanks again
> Bruce




From shaheryarkh at googlemail.com  Mon Mar  8 07:16:34 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 8 Mar 2010 20:16:34 +0500
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <b16156851003080716l6e6d5b0s47bc57724ab14eda@mail.gmail.com>

I have CISCO AS 5350 running with Freeswitch in production environment.
Seems to work perfect.

Thank you.


On Mon, Mar 8, 2010 at 8:08 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From yehavi.bourvine at gmail.com  Mon Mar  8 07:18:45 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 8 Mar 2010 17:18:45 +0200
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <d361ca7b1003080718l6aaf9db7t6f72707ac17cded9@mail.gmail.com>

What help do you need? What Cisco do you have?

                __Yehavi:

 2010/3/8 srinivasula reddy <srinivas.ksvreddy at gmail.com>

>  Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From srinivas.ksvreddy at gmail.com  Mon Mar  8 07:23:35 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 8 Mar 2010 20:53:35 +0530
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
Message-ID: <f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>

HI,

how do i configure freeswitch gateway to communicate to cisco router, i am
using cisco2921.

Thanks
Srinivas
On Mon, Mar 8, 2010 at 8:38 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> can any body tried to connect cisco router from Freeswitch, please help me.
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From brian at freeswitch.org  Mon Mar  8 07:33:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:33:34 -0600
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
Message-ID: <9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>

You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.

/b

On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:

> 
> HI,
>  
> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
>  
> Thanks
> Srinivas




From kristian.kielhofner at gmail.com  Mon Mar  8 07:40:50 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Mar 2010 10:40:50 -0500
Subject: [Freeswitch-users] Attrafax
Message-ID: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>

Disclaimers:

- I haven't looked at it in detail (at all, really)
- It is GPL2 (although perhaps Zoa and co can budge on this)

http://www.venturevoip.com/news.php?rssid=2364

Any thoughts?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From vizentini at hotmail.com  Mon Mar  8 07:42:43 2010
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Mon, 8 Mar 2010 15:42:43 +0000
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>,
	<4B9409FF.5060608@gmail.com>,
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>,
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>,
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>,
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>,
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>,
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
Message-ID: <SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>



Hi
"As per FS wiki we can only return one XML extension upon FS request. "
Could you please tell me where did you find that info? I do return several extensions by request and FS works as expect (Am I wrong doing that?)
"Now we need to return a set of extensions with same number
condition and context but with different second condition so that FS
can pick up the right one. 
Is it possible at all? And if yes how response XML with multiple extensions should look like?"
Yes, It seems to be possible (Am I wrong?) but notice that you are giving the  same extension name for all your extensions:
 <extension name="test9">

I think this can mislead dialplan_hunt in mod_dialplan_xml...Give each extension a unique name, and FS will parse/regex all them for you... 

All the best,Paulo
> From: brian at freeswitch.org
> Date: Sun, 7 Mar 2010 20:17:19 -0600
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
> 
> Sounds like you zigged where you should have zagged.
> 
> /b
> 
> On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> 
> > :-))
> >  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> > So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From brian at freeswitch.org  Mon Mar  8 07:47:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 09:47:44 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
Message-ID: <007EE533-8154-4B00-AEFD-3857E684E9EB@freeswitch.org>

We have SpanDSP and its LGPL.  

/b

On Mar 8, 2010, at 9:40 AM, Kristian Kielhofner wrote:

> Disclaimers:
> 
> - I haven't looked at it in detail (at all, really)
> - It is GPL2 (although perhaps Zoa and co can budge on this)
> 
> http://www.venturevoip.com/news.php?rssid=2364
> 
> Any thoughts?
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From steveu at coppice.org  Mon Mar  8 07:57:23 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 08 Mar 2010 23:57:23 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
Message-ID: <4B951E63.6060200@coppice.org>

On 03/08/2010 11:40 PM, Kristian Kielhofner wrote:
> Disclaimers:
>
> - I haven't looked at it in detail (at all, really)
> - It is GPL2 (although perhaps Zoa and co can budge on this)
>
> http://www.venturevoip.com/news.php?rssid=2364
>
> Any thoughts?
>    
We already have a better FAX engine than that. What we struggle to find 
is the time to properly integrate it with FS.

Steve




From m.sobkow at marketelsystems.com  Mon Mar  8 08:22:34 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Mar 2010 10:22:34 -0600
Subject: [Freeswitch-users] Eavesdrop
Message-ID: <4B95244A.8030808@marketelsystems.com>

What module contains the "eavesdrop" command?  I'm getting an error 
command not found from Erlang when I try to do a pbx:api( eavesdrop, 
ExtensionUUID ).

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




From kristian.kielhofner at gmail.com  Mon Mar  8 08:24:47 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Mon, 8 Mar 2010 11:24:47 -0500
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4B951E63.6060200@coppice.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
Message-ID: <4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>

On Mon, Mar 8, 2010 at 10:57 AM, Steve Underwood <steveu at coppice.org> wrote:
> We already have a better FAX engine than that. What we struggle to find
> is the time to properly integrate it with FS.
>
> Steve

  Thank you, that's what I was expecting.

  Without going into grueling detail it isn't clear why T.38 gateway
functionality (for example) is missing in FreeSWITCH.  I know SpanDSP
is available and most likely has this ability.  I just wasn't sure why
it's missing in FS.  This is also a mere curiosity for me; I don't
have much interest in fax functionality and I don't follow it too
closely.

  Thanks again for the response and explanation.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From rupa at rupa.com  Mon Mar  8 08:55:05 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 8 Mar 2010 10:55:05 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B95244A.8030808@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
Message-ID: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>

application,eavesdrop,mod_dptools,/opt/freeswitch/mod/mod_dptools.so

It is an app, not an API.

On Mon, Mar 8, 2010 at 10:22 AM, Mark Sobkow
<m.sobkow at marketelsystems.com>wrote:

> What module contains the "eavesdrop" command?  I'm getting an error
> command not found from Erlang when I try to do a pbx:api( eavesdrop,
> ExtensionUUID ).
>
> --
> Mark Sobkow
> Senior Developer
> MarkeTel Multi-Line Dialing Systems LTD.
> 428 Victoria Ave
> Regina, SK S4N-0P6
> Toll-Free: 800-289-8616-X533
> Local: 306-359-6893-X533
> Fax: 306-359-6879
> Email: m.sobkow at marketelsystems.com
> Web: http://www.marketelsystems.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From mike at van.lammeren.net  Mon Mar  8 09:00:37 2010
From: mike at van.lammeren.net (Mike van Lammeren)
Date: Mon, 8 Mar 2010 12:00:37 -0500
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
	<23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
Message-ID: <5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>

Actually, I don't believe that does work. At least, not on my system. Trying
to access argv[0] just produces a lua error about "concatenating a nil
value."

On Fri, Mar 5, 2010 at 10:28 PM, Seven Du <dujinfang at gmail.com> wrote:

> and also you can get the script name from argv[0] if you name you
> script in different names
>
> 2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
> > In case anyone's interested, I just found something cool. You can pass
> > parameters to the Lua script from the lua.conf.xml file, like so:
> >     <param name="startup-script" value="temp.lua script1"/>
> >     <param name="startup-script" value="temp.lua script2"/>
> >     <param name="startup-script" value="temp.lua script3"/>
> > I can see that being useful.
> > Mike van Lammeren
> >
> > On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren <mike at van.lammeren.net
> >
> > wrote:
> >>
> >> Thanks so much, everyone! Awesome advice, as usual!
> >> Due to a deadline situation, I'm going with the Lua script idea.
> However,
> >> if it gives us any trouble after the looming deadline, then I will
> >> definitely go the C route.
> >> Thanks again!
> >> Mike van Lammeren
> >>
> >> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org>
> wrote:
> >>>
> >>> Write it in C and use the FreeSWITCH task scheduler API.
> >>>
> >>> /b
> >>>
> >>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
> >>>
> >>> > All the same, I'm still open to suggestions on better ways to achieve
> >>> > my ultimate goals!
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon Mar  8 09:07:03 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 12:07:03 -0500
Subject: [Freeswitch-users] FScomm
References: <157FDB96FB88476FBAEC69C1309A0A5F@MOBILEE1705><COL117-W5563292C886257AFC56548B9420@phx.gbl><D64783281177440B9094B8181E55EFA7@MOBILEE1705><COL117-W429F3BED9F781A58264238B9420@phx.gbl><11F8492DFC2546598E6FDC3E538301CD@MOBILEE1705><BAA7D201-6E09-4EDE-BBA0-FDFA64E38391@jerris.com><D262917F3F4943D49370D0CC015A7582@MOBILEE1705><ebbc2d551003022200n73e56fd3nb4dd65b43407c36@mail.gmail.com><13F83F4CC35047199F95ED9EA80149BE@MOBILEE1705>
	<ebbc2d551003072015t7537eae4g6564c1286fb1fd3b@mail.gmail.com>
Message-ID: <40DCBC09D015499EB5EC5E2B292A3515@MOBILEE1705>

sure, I send to you the connection info now

Regards

  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 07, 2010 11:15 PM
  Subject: Re: [Freeswitch-users] FScomm


  If I can access your system over SSH I can probably figure this out. Is this possible?


  Jo?o Mesquita



  On Wed, Mar 3, 2010 at 3:26 AM, Madovsky <infos at madovsky.org> wrote:

    Hi Joao,

    thanks for your answer.
    I have a standard RPM QT last update for Fedora10 64bits.
    so I really don't know what path FScomm needs

    Regards

    Franck
      ----- Original Message ----- 
      From: Jo?o Mesquita 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 03, 2010 1:00 AM
      Subject: Re: [Freeswitch-users] FScomm


      This is definitely a problem with your Qt installation. You need to be able to find the libs to link it, otherwise, unresolved symbols everywhere! 


      Regards,
      Jo?o Mesquita
      FSComm Developer



      On Wed, Feb 24, 2010 at 2:42 PM, Madovsky <infos at madovsky.org> wrote:


        ----- Original Message -----
        From: "Michael Jerris" <mike at jerris.com>

        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Wednesday, February 24, 2010 3:40 AM

        Subject: Re: [Freeswitch-users] FScomm



        On Feb 23, 2010, at 6:18 PM, Madovsky wrote:

        >
        > ----- Original Message -----
        > From: Jeff Lenk
        > To: freeswitch-users at lists.freeswitch.org
        > Sent: Tuesday, February 23, 2010 5:05 PM
        > Subject: Re: [Freeswitch-users] FScomm
        >
        >
        > http://wiki.freeswitch.org/wiki/FSComm#Linux
        >
        > you must run those from the FSComm directory
        >

        ?.

        >
        > It's what I did,
        > but from FS trunk, inside fscomm directory,
        > there s only
        >
        > account.cpp  conf                fshost.h        mainwindow.ui
        > resources.qrc
        > account.h    FSComm.2008.vcproj  main.cpp        mod_qsettings
        > call.cpp     FSComm.pro          mainwindow.cpp  preferences
        > call.h       fshost.cpp          mainwindow.h    resources
        >


        Read those installation instructions again and do them step by step, you
        skipped one.
        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org



        Ok now I have

        [root at node250 fscomm]# qmake
        WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
        in SOURCES
        WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
        HEADERS
        WARNING: Found potential symbol conflict of prefdialog.cpp
        (preferences/prefdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of prefdialog.h
        (preferences/prefdialog.h) in HEADERS
        WARNING: Found potential symbol conflict of accountdialog.cpp
        (preferences/accountdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of accountdialog.h
        (preferences/accountdialog.h) in HEADERS

        [root at node250 fscomm]# qmake
        WARNING: Found potential symbol conflict of mainwindow.cpp (mainwindow.cpp)
        in SOURCES
        WARNING: Found potential symbol conflict of mainwindow.h (mainwindow.h) in
        HEADERS
        WARNING: Found potential symbol conflict of prefdialog.cpp
        (preferences/prefdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of prefdialog.h
        (preferences/prefdialog.h) in HEADERS
        WARNING: Found potential symbol conflict of accountdialog.cpp
        (preferences/accountdialog.cpp) in SOURCES
        WARNING: Found potential symbol conflict of accountdialog.h
        (preferences/accountdialog.h) in HEADERS
        [root at node250 fscomm]# make
        Makefile:278: warning: overriding commands for target `prefdialog.o'
        Makefile:215: warning: ignoring old commands for target `prefdialog.o'
        Makefile:285: warning: overriding commands for target `accountdialog.o'
        Makefile:234: warning: ignoring old commands for target `accountdialog.o'
        Makefile:320: warning: overriding commands for target `moc_prefdialog.o'
        Makefile:298: warning: ignoring old commands for target `moc_prefdialog.o'
        Makefile:323: warning: overriding commands for target `moc_accountdialog.o'
        Makefile:307: warning: ignoring old commands for target
        `moc_accountdialog.o'
        Makefile:347: warning: overriding commands for target `moc_mainwindow.cpp'
        Makefile:326: warning: ignoring old commands for target `moc_mainwindow.cpp'
        Makefile:350: warning: overriding commands for target
        `preferences/moc_prefdialog.cpp'
        Makefile:332: warning: ignoring old commands for target
        `preferences/moc_prefdialog.cpp'
        Makefile:353: warning: overriding commands for target
        `preferences/moc_accountdialog.cpp'
        Makefile:341: warning: ignoring old commands for target
        `preferences/moc_accountdialog.cpp'
        g++ -c -pipe -Wall -W -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
         -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic  -DQT_NO_DEBUG
         -DQT_SHARED -DQT_TABLET_SUPPORT -DQT_THREAD_SUPPORT -I/usr/lib64/qt-3.3/mkspecs/default
         -I. -I../src/include -I../libs/apr/include -I../libs/libteletone/src -I/usr/lib64/qt-3.3/include
         -o main.o main.cpp
        main.cpp:31:25: error: QSplashScreen: No such file or directory
        In file included from main.cpp:32:
        mainwindow.h:34:23: error: QMainWindow: No such file or directory
        mainwindow.h:35:28: error: QTableWidgetItem: No such file or directory
        mainwindow.h:36:25: error: QSignalMapper: No such file or directory
        mainwindow.h:37:27: error: QSystemTrayIcon: No such file or directory
        In file included from mainwindow.h:39,
                        from main.cpp:32:
        ./fshost.h:32:19: error: QThread: No such file or directory
        ./fshost.h:33:17: error: QHash: No such file or directory
        ./fshost.h:34:26: error: QSharedPointer: No such file or directory
        In file included from ./fshost.h:36,
                        from mainwindow.h:39,
                        from main.cpp:32:
        ./call.h:32:18: error: QtCore: No such file or directory
        ./call.h:33:19: error: QString: No such file or directory
        In file included from mainwindow.h:42,
                        from main.cpp:32:
        preferences/prefdialog.h:4:19: error: QDialog: No such file or directory
        preferences/prefdialog.h:5:24: error: QDomDocument: No such file or
        directory
        preferences/prefdialog.h:6:21: error: QSettings: No such file or directory
        In file included from ./fshost.h:37,
                        from mainwindow.h:39,
                        from main.cpp:32:
        ./account.h:18: error: expected constructor, destructor, or type conversion
        before ?static?
        In file included from mainwindow.h:39,
                        from main.cpp:32:
        ./fshost.h:40: error: invalid use of incomplete type ?struct QThread?
        /usr/include/QtCore/qobject.h:68: error: forward declaration of ?struct
        QThread?
        ./fshost.h:46: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:46: error: expected ?;? before ?<? token
        ./fshost.h:47: error: expected `;' before ?QSharedPointer?
        ./fshost.h:47: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:47: error: expected ?;? before ?<? token
        ./fshost.h:48: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:48: error: template argument 1 is invalid
        ./fshost.h:48: error: expected unqualified-id before ?>? token
        ./fshost.h:49: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:49: error: expected ?;? before ?<? token
        ./fshost.h:50: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:50: error: expected ?;? before ?<? token
        ./fshost.h:51: error: ISO C++ forbids declaration of ?QSharedPointer? with
        no type
        ./fshost.h:51: error: expected ?;? before ?<? token
        ./fshost.h:52: error: ?QSharedPointer? has not been declared
        ./fshost.h:52: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:60: error: ?QSharedPointer? has not been declared
        ./fshost.h:60: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:61: error: ?QSharedPointer? has not been declared
        ./fshost.h:61: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:62: error: ?QSharedPointer? has not been declared
        ./fshost.h:62: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:63: error: ?QSharedPointer? has not been declared
        ./fshost.h:63: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:64: error: ?QSharedPointer? has not been declared
        ./fshost.h:64: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:65: error: ?QSharedPointer? has not been declared
        ./fshost.h:65: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:66: error: ?QSharedPointer? has not been declared
        ./fshost.h:66: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:67: error: ?QSharedPointer? has not been declared
        ./fshost.h:67: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:71: error: ?QSharedPointer? has not been declared
        ./fshost.h:71: error: expected ?,? or ?...? before ?<? token
        ./fshost.h:78: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:78: error: template argument 2 is invalid
        ./fshost.h:78: error: expected unqualified-id before ?>? token
        ./fshost.h:79: error: ?QSharedPointer? was not declared in this scope
        ./fshost.h:79: error: template argument 2 is invalid
        ./fshost.h:79: error: expected unqualified-id before ?>? token
        ./fshost.h:80: error: field ?_bleg_uuids? has incomplete type
        In file included from mainwindow.h:42,
                        from main.cpp:32:
        preferences/prefdialog.h:17: error: invalid use of incomplete type ?struct
        QDialog?
        /usr/include/QtGui/qwindowdefs.h:57: error: forward declaration of ?struct
        QDialog?
        preferences/prefdialog.h:31: error: ISO C++ forbids declaration of
        ?QSettings? with no type
        preferences/prefdialog.h:31: error: expected ?;? before ?*? token
        In file included from main.cpp:32:
        mainwindow.h:48: error: expected class-name before ?{? token
        mainwindow.h:65: error: ?QTableWidgetItem? has not been declared
        mainwindow.h:71: error: ?QSharedPointer? has not been declared
        mainwindow.h:71: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:72: error: ?QSharedPointer? has not been declared
        mainwindow.h:72: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:73: error: ?QSharedPointer? has not been declared
        mainwindow.h:73: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:74: error: ?QSharedPointer? has not been declared
        mainwindow.h:74: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:75: error: ?QSharedPointer? has not been declared
        mainwindow.h:75: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:78: error: ?QSharedPointer? has not been declared
        mainwindow.h:78: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:79: error: ?QSharedPointer? has not been declared
        mainwindow.h:79: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:80: error: ?QSharedPointer? has not been declared
        mainwindow.h:80: error: expected ?,? or ?...? before ?<? token
        mainwindow.h:81: error: ?QSystemTrayIcon? has not been declared
        mainwindow.h:81: error: expected ?,? or ?...? before ?reason?
        mainwindow.h:86: error: ISO C++ forbids declaration of ?QSignalMapper? with
        no type
        mainwindow.h:86: error: expected ?;? before ?*? token
        mainwindow.h:88: error: ISO C++ forbids declaration of ?QSystemTrayIcon?
        with no type
        mainwindow.h:88: error: expected ?;? before ?*? token
        main.cpp: In function ?int main(int, char**)?:
        main.cpp:41: error: variable ?QPixmap image? has initializer but incomplete
        type
        main.cpp:42: error: ?QSplashScreen? was not declared in this scope
        main.cpp:42: error: ?splash? was not declared in this scope
        main.cpp:42: error: expected type-specifier before ?QSplashScreen?
        main.cpp:42: error: expected `;' before ?QSplashScreen?
        main.cpp:48: error: no matching function for call to
        ?QObject::connect(FSHost*, const char [9], MainWindow*, const char [8])?
        /usr/include/QtCore/qobject.h:202: note: candidates are: static bool
        QObject::connect(const QObject*, const char*, const QObject*, const char*,
        Qt::ConnectionType)
        /usr/include/QtCore/qobject.h:307: note:                 bool
        QObject::connect(const QObject*, const char*, const char*,
        Qt::ConnectionType) const
        main.cpp:49: error: ?class FSHost? has no member named ?start?
        main.cpp:41: warning: unused variable ?image?
        ./fshost.h: At global scope:
        ./fshost.h:90: warning: ?void eventHandlerCallback(switch_event_t*)? defined
        but not used
        make: *** [main.o] Error 1


        Any idea ?

        Thx




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





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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jbrucehopkins at gmail.com  Mon Mar  8 09:12:42 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 8 Mar 2010 17:12:42 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
Message-ID: <d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>

Hi again,

Thanks for the help so far.  Unfotrunately I must still be doing something
wrong here as I am still having difficulty, and still have the same problem.

I updated to build 16938 by means of "make current"

I'm not able to use UPNP or NATPMP so changed the port forwarding to use
5080 instead of 5090.

I got rid of the doublenat profile in sip_profiles, though I had to retain
an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml
in order to give freeswitch the dns name of the server as a domain for the
remote softphone to register on.  I left the group name in this entry the
same as inthe default entry, so that the remote phone could register on the
same extension numbers (100, etc) as in the default build.

I still find that, if I initiate a call from the local (on same LAN as
freeswitch) phone to the remote phone, I get the message on the CLI:

[ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type
[user] cause: [USER_NOT_REGISTERED]

One possibly unrelated aside, I also found I needed to uncomment <param
name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in
the case of a call initiated by the remote phone being hung up by the local
phone, freeswitch sent the BYE to the private IP of the remote phone, rather
than its public ip - meaning that the remote phone didn't receive the BYE.

Any further ideas where I am going wrong here please?

thanks again in advance
Bruce



On 8 March 2010 15:15, Brian West <brian at freeswitch.org> wrote:

> Well sounds like you need to sync in your configs too.
>
> /b
>
> On Mar 8, 2010, at 9:03 AM, Bruce Hopkins wrote:
>
> > Many thanks for getting back to me with help so quickly Brian.
> >
> > I was using 1.0.5-20100302-0400 so will update to trunk and do as you
> suggest.
> >
> > thanks again
> > Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Sat Mar  6 21:02:42 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sun, 7 Mar 2010 16:02:42 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
Message-ID: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>

Hi,
I am trying to create a mod_managed API application that takes 2 phone
numbers as params.  These numbers should then be used to make 2 calls and to
bridge them. The issue i am having is working out how to place the first
call leg from in mod_managed.  All the methods i have found are for the
second leg and require you to pass in the first call leg.

I know there are easier way to do this out of the managed code but i want it
in the managed code as this is just a proof of concept to prove we can do a
callback like system via managed code to be hooked into other external
systems.
In wiki http://wiki.freeswitch.org/wiki/Session there is this example

s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
while (s.ready()) {
// The call has been answered
}

This seems to be exactly what i am looking for but seems to be missing from
mod_managed. I am hoping someone can tell me how to do something similar in
mod_managed since even if it is not part of the managed code mod_managed is
meant to have the native access as well so i would assume it would be
possible.

Thanks in advance

Chris
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From its.solution at cox.net  Mon Mar  8 07:16:05 2010
From: its.solution at cox.net (ITS Solution)
Date: Mon, 08 Mar 2010 10:16:05 -0500
Subject: [Freeswitch-users] Need help on dingaling module
Message-ID: <C7BA7EE5.3FB3%its.solution@cox.net>

Hi all,

I recently installed FS version 16983 with dingaling module enabled. I am
using the client configuration. My system is Linux Ubuntu 8.04 hardy running
on Xen VM. When I set up a call between a phone and a google talk, the audio
from google talk is all noise. The audio from phone to google talk sounds
good though. Before I did the upgrade, I was running on version  16892. In
that release, I can hear audio on both paths. The audio on the path from
google talk to FS results in about 3-5 seconds delay.

Does anybody experience the similar issue or can help me to point out where
could be the problem?

Thanks.

Chaofeng
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From brian at freeswitch.org  Mon Mar  8 09:21:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 11:21:48 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
Message-ID: <8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>

ok you aren't catching one key thing here.. you no longer need two profiles.

/b

On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:

> Hi again,
> 
> Thanks for the help so far.  Unfotrunately I must still be doing something wrong here as I am still having difficulty, and still have the same problem.
> 
> I updated to build 16938 by means of "make current"
> 
> I'm not able to use UPNP or NATPMP so changed the port forwarding to use 5080 instead of 5090.
> 
> I got rid of the doublenat profile in sip_profiles, though I had to retain an entry in the directory /usr/local/freeeswitch/conf/directory/ext_dns.xml in order to give freeswitch the dns name of the server as a domain for the remote softphone to register on.  I left the group name in this entry the same as inthe default entry, so that the remote phone could register on the same extension numbers (100, etc) as in the default build.
> 
> I still find that, if I initiate a call from the local (on same LAN as freeswitch) phone to the remote phone, I get the message on the CLI:
> 
> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
> 
> One possibly unrelated aside, I also found I needed to uncomment <param name="aggressive-nat-detection" value="true"/> in external.xml, otherwise in the case of a call initiated by the remote phone being hung up by the local phone, freeswitch sent the BYE to the private IP of the remote phone, rather than its public ip - meaning that the remote phone didn't receive the BYE.
> 
> Any further ideas where I am going wrong here please?
> 
> thanks again in advance
> Bruce




From msc at freeswitch.org  Mon Mar  8 09:22:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 09:22:01 -0800
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
Message-ID: <87f2f3b91003080922o34d5495foc9641bfc3357989@mail.gmail.com>

On Mon, Mar 8, 2010 at 8:55 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> application,eavesdrop,mod_dptools,/opt/freeswitch/mod/mod_dptools.so
>
> It is an app, not an API.
>

I other words use session:execute("eavesdrop") and not pbx:api("eavesdrop")
-MC
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From t.mahe at telemaque.fr  Mon Mar  8 09:23:55 2010
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 08 Mar 2010 18:23:55 +0100
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
Message-ID: <4B9532AB.5030407@telemaque.fr>

+1 with Brian,

Just as I have 5 spare minutes and to be kind:


* E1 controller config:

controller E1 0/0/0
 framing NO-CRC4
 pri-group timeslots 1-31
 description E1_1
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn not-end-to-end 64
 isdn incoming-voice voice
 isdn bchan-number-order descending round-robin
 no cdp enable
!
voice-port 0/0/0:15
 no comfort-noise
 cptone FR
 timeouts call-disconnect 1
 timeouts wait-release 1
!

* Dial-peers ( SIP to E1 and E1 to SIP):

dial-peer voice 100 pots
 description SIP_TO_E1
 destination-pattern 0........
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 progress_ind disconnect enable 8
 no digit-strip
 port 0/0/0:15
!
dial-peer voice 500 voip
 description E1_TO_SIP
 destination-pattern ....
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:FREESWITCH_IP
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!


Remember to adapt this config to your needs ( there are mandatory
missing parts, but at least you have some config options to look for).

Regards,

Gled.

Brian West a ?crit :
> You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.
> 
> /b
> 
> On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> 
>> HI,
>>  
>> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
>>  
>> Thanks
>> Srinivas
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From paul.gore.j at gmail.com  Mon Mar  8 09:22:37 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 8 Mar 2010 12:22:37 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<4B9409FF.5060608@gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
Message-ID: <da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>

It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will
only need to send one extension back..."
In my real dial plan I do use unique names for the extensions, but as I
mentioned FS only executes first extension from the response. If I place
same extensions in a static XML they work as  expected.
I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you
post a sample response you generate, please?


On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com>wrote:

>
> Hi
>
> "As per FS wiki we can only return one XML extension upon FS request. "
>
> Could you please tell me where did you find that info? I do return several
> extensions by request and FS works as expect (Am I wrong doing that?)
>
> "Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?"
>
> Yes, It seems to be possible (Am I wrong?) but notice that you are giving
> the  same extension name for all your extensions:
>
>  <extension name="test9">
>
>
> I think this can mislead dialplan_hunt in mod_dialplan_xml...
> Give each extension a unique name, and FS will parse/regex all them for
> you...
>
>
> All the best,
> Paulo
>
> > From: brian at freeswitch.org
> > Date: Sun, 7 Mar 2010 20:17:19 -0600
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] XML curl and multiple extension in
> dialplan
>
> >
> > Sounds like you zigged where you should have zagged.
> >
> > /b
> >
> > On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> >
> > > :-))
> > > I used it a lot. Unfortunately we do not store did number as a field,
> as I said we store complete extension XML, with extension number and context
> as a key.
> > > So essentially I will retrieve a bunch of extension XMLs which I need
> to parse and run against regex matcher to find out the one, which is what
> exactly FS is doing when XML dial plan is defined in a static file.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 09:32:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 11:32:05 -0600
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <4B9532AB.5030407@telemaque.fr>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr>
Message-ID: <23FE228D-9CA2-4858-8A94-5AD5FD770FA8@freeswitch.org>

Thank you for taking the time to help out... maybe a wiki page can be created on this moving forward.  I wasn't in any way trying to be mean or disrespectful.  I'm clearly trying to state that we can't possibly help everyone with everything at some point it just starts to get overwhelming to try to help everyone.  I thank everyone that are starting to step up and help people out on the list.  Lets keep that up it helps greatly.

Thanks,
Brian

On Mar 8, 2010, at 11:23 AM, Tristan Mah? wrote:

> +1 with Brian,
> 
> Just as I have 5 spare minutes and to be kind:




From will.traenkle at yahoo.com  Mon Mar  8 10:05:24 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 10:05:24 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <756904.84769.qm@web57613.mail.re1.yahoo.com>

I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

My hunt group code is below:


session.answer();
var domain_name = session.getVariable("domain_name");
var extension = '7000';
var result;
var timeoutpin = 7500;

var objdate = new Date();
var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours
var adjustoperator = "-"; //+ or -
if (adjustoperator == "-") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
if (adjustoperator == "+") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
var Hours = objdate2.getHours();
var Mins = objdate2.getMinutes();
var Seconds = objdate2.getSeconds();
var Month = objdate2.getMonth() + 1;
var Date = objdate2.getDate();
var Year = objdate2.getYear()
var Day = objdate2.getDay()+1;
var exit = false;

 function get_sofia_contact(extension,domain_name, profile){
        if (profile == "auto") {
                        profile = "internal";
                        session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                        sofia_contact = session.getVariable("sofia_contact_"+extension);

                        if (sofia_contact == "error/user_not_registered") {
                                profile = "external";
                                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                                sofia_contact = session.getVariable("sofia_contact_"+extension);
                        }
        }
        else {
                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                sofia_contact = session.getVariable("sofia_contact_"+extension);
        }
        console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" );
        return sofia_contact;
 }


 function mycb( session, type, obj, arg ) {
    try {
        if ( type == "dtmf" ) {
          console_log( "info", "digit: "+obj.digit+"\n" );
          if ( obj.digit == "#" ) {
            //console_log( "info", "detected pound sign.\n" );
            exit = true;
            return( false );
          }

          dtmf.digits += obj.digit;

          if ( dtmf.digits.length >= digitmaxlength ) {
            exit = true;
            return( false );
          }
        }
    } catch (e) {
        console_log( "err", e+"\n" );
    }
    return( true );
 } //end function mycb


dialed_extension = session.getVariable("dialed_extension");
domain_name = session.getVariable("domain_name");
domain = session.getVariable("domain");
us_ring = session.getVariable("us-ring");
caller_id_name = session.getVariable("caller_id_name");
caller_id_number = session.getVariable("caller_id_number");
effective_caller_id_name = session.getVariable("effective_caller_id_name");
effective_caller_id_number = session.getVariable("effective_caller_id_number");
outbound_caller_id_name = session.getVariable("outbound_caller_id_name");
outbound_caller_id_number = session.getVariable("outbound_caller_id_number");
session.execute("set", "ringback=${hold_music}");          //set to ringtone
session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone
session.execute("set", "call_timeout=30");
session.execute("export", "call_timeout=30");
session.execute("set", "continue_on_fail=true");
session.execute("set", "hangup_after_bridge=true");

//console_log( "info", "dialed extension:"+dialed_extension+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "us_ring: "+us_ring+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "domain_name: "+domain_name+".\n" );

//console_log( "info", "action call now don't wait for dtmf\n" );
if ( session.ready() ) {
        session.answer();

        sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto");

        session.execute("bridge", sofia_contact_1000);
        session.execute("bridge", "sofia/gateway/flowroute.com/16194548924");

        //timeout
        if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){
                session.execute("info", "");
                session.execute("transfer", "*991000");
        }

        //clear variables
        dialed_extension = "";
        new_extension = "";
        domain_name = "";
        domain = "";
} //end if session.ready


Thanks.

-Will


      
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From lloyd.aloysius at gmail.com  Mon Mar  8 10:23:54 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 13:23:54 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer Help
Message-ID: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>

Hi All,

I would like to setup a dial plan, Dial *99[extension number]  send a call
directly to voicemail.

 I setup the following Dial Plan . But when I transfer the call
*99[extension number] the call failed.

    <extension name="voicemail_transfer">
      <condition field="destination_number" expression="^(\*9920[0-9])$">
  <action application="answer"/>
  <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
  </condition>
    </extension>
But when I transfer the call *99[extension number] the call failed.My
extensions range 201 - 209

Please let me know where the mistake is.

Thanks
Lloyd
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From vizentini at hotmail.com  Mon Mar  8 10:31:39 2010
From: vizentini at hotmail.com (Paulo Vicentini)
Date: Mon, 8 Mar 2010 18:31:39 +0000
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>,
	<4B9409FF.5060608@gmail.com>,
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>,
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>,
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>,
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>,
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>,
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>,
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>,
	<da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
Message-ID: <SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>


Hello,
FreeSWITCH version: 1.0.trunk (16870M)
I have replied FS even with extensions you have emailed here (in my case I have data="sofia/gateway/myGW/888 at conference.freeswitch.org/888") - it works
You may need to give a look at "CS_ROUTING" state.

best,PauloDate: Mon, 8 Mar 2010 12:22:37 -0500
From: paul.gore.j at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan

It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will only need to send one extension back..."
In my real dial plan I do use unique names for the extensions, but as I mentioned FS only executes first extension from the response. If I place same extensions in a static XML they work as  expected.
I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you post a sample response you generate, please?


On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com> wrote:







Hi
"As per FS wiki we can only return one XML extension upon FS request. "
Could you please tell me where did you find that info? I do return several extensions by request and FS works as expect (Am I wrong doing that?)

"Now we need to return a set of extensions with same number
condition and context but with different second condition so that FS
can pick up the right one. 
Is it possible at all? And if yes how response XML with multiple extensions should look like?"
Yes, It seems to be possible (Am I wrong?) but notice that you are giving the  same extension name for all your extensions:

 <extension name="test9">

I think this can mislead dialplan_hunt in mod_dialplan_xml...Give each extension a unique name, and FS will parse/regex all them for you... 

All the best,
Paulo
> From: brian at freeswitch.org
> Date: Sun, 7 Mar 2010 20:17:19 -0600
> To: freeswitch-users at lists.freeswitch.org

> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
> 
> Sounds like you zigged where you should have zagged.
> 
> /b
> 
> On Mar 7, 2010, at 8:03 PM, paul gore wrote:

> 
> > :-))
> >  I used it a lot. Unfortunately we do not store did number as a field, as I said we store complete extension XML, with extension number and context as a key.
> > So essentially I will retrieve a bunch of extension XMLs which I need to parse and run against regex matcher to find out the one, which is what exactly FS is doing when XML dial plan is defined in a static file.

> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org

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

> http://www.freeswitch.org
 		 	   		  
Hotmail: Powerful Free email with security by Microsoft. Get it now.


_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

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_________________________________________________________________
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From gavin.henry at gmail.com  Mon Mar  8 10:33:27 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 18:33:27 +0000
Subject: [Freeswitch-users] Virtualized FreeSWITCH
In-Reply-To: <70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>
References: <7454A296C7EDE34EA57199FAA401E2F11C63F2D970@VMBX113.ihostexchange.net>
	<7454A296C7EDE34EA57199FAA401E2F11C63F2DA66@VMBX113.ihostexchange.net>
	<7b197bef1003031300j756aa0ddv777406fc3224da9c@mail.gmail.com>
	<25BD7E1E-B846-43AA-B587-1FCCE5667E79@gmail.com>
	<7b197bef1003031315tfe57e3fs98cfcfd9edf92939@mail.gmail.com>
	<23f91031003031704ob68528g37f64f9b0ee7a2bd@mail.gmail.com>
	<9cb0e15e1003040655p500dae55kc22de012c270aedd@mail.gmail.com>
	<e15d328c1003071834v1be38b48p4028e9add2b748f8@mail.gmail.com>
	<13ca621c1003080134n7e2a5d14kb5c1744ddc85d665@mail.gmail.com>
	<70064BF8-B34E-4320-8B77-42CF82E958A1@gmail.com>
Message-ID: <13ca621c1003081033n5ab07ea6lf3233194d4027fb3@mail.gmail.com>

 On 8 March 2010 11:30, Jay Binks <jaybinks at gmail.com> wrote:
> The qn here is do you bypass media or do you do complete topology
> hiding ??
>
> Sip only is kind of a given , ?it gets harder if the virtualized pbx
> does media.

The pbx does the media here and then on to our VoIP provider platform.



From brian at freeswitch.org  Mon Mar  8 10:33:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 12:33:26 -0600
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <0419C3ED-7F68-4299-8170-39C376FE3CC0@freeswitch.org>

That looks correct can you show me console output?

/b

On Mar 8, 2010, at 12:23 PM, Aloysius Lloyd wrote:

> Hi All,
> 
> I would like to setup a dial plan, Dial *99[extension number]  send a call directly to voicemail.
> 
>  I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed.
> 
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
> 	  <action application="answer"/>
> 	  <action application="sleep" data="1000"/>
> 	  <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
> 	  </condition>
>     </extension>
> 	
> But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209
> 
> Please let me know where the mistake is.
> 
> Thanks
> Lloyd

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From chris.chen2004 at gmail.com  Mon Mar  8 10:34:57 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 13:34:57 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <507898381003081034t13752159y87c2deb138fbb5f6@mail.gmail.com>

Lloyd, you didn't define ${dialed_extension}, before you call the voicemail
application in your dialplan:
 <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>

Please fix that.

Chris

On Mon, Mar 8, 2010 at 1:23 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
>
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
>   <action application="answer"/>
>   <action application="sleep" data="1000"/>
>   <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>   </condition>
>     </extension>
> But when I transfer the call *99[extension number] the call failed.My
> extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rob4manhere at gmail.com  Mon Mar  8 10:34:15 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Mon, 8 Mar 2010 12:34:15 -0600
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <95DBAAA1-EB57-4440-8C7B-618053730DB5@gmail.com>

Much easier to see what's going on with logs.  Post the snippet of a  
test call from your freeswitch logs in debug to http://pastebin.freeswitch.org/ 
  and share the link.


On Mar 8, 2010, at 12:23 PM, Aloysius Lloyd wrote:

> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send  
> a call directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call  
> *99[extension number] the call failed.
>
>     <extension name="voicemail_transfer">
>       <condition field="destination_number"  
> expression="^(\*9920[0-9])$">
> 	  <action application="answer"/>
> 	  <action application="sleep" data="1000"/>
> 	  <action application="voicemail" data="default ${domain_name} $ 
> {dialed_extension}"/>
> 	  </condition>
>     </extension>
> 	
> But when I transfer the call *99[extension number] the call  
> failed.My extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From frank at carmickle.com  Mon Mar  8 10:40:25 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 8 Mar 2010 13:40:25 -0500
Subject: [Freeswitch-users] hunt group - play music when trying	external
	numbers
In-Reply-To: <756904.84769.qm@web57613.mail.re1.yahoo.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
Message-ID: <20100308184025.GL18427@base.carmickle.com>

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC



From frank at carmickle.com  Mon Mar  8 10:45:17 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Mon, 8 Mar 2010 13:45:17 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail
	Transfer	Help
In-Reply-To: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <20100308184517.GM18427@base.carmickle.com>

Hello

On Mon, Mar 08, Aloysius Lloyd wrote:
> Hi All,
> 
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
> 
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
> 
>     <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">

Should be

<condition field="destination_number" expression="^\*99(20[0-9])$">

>   <action application="answer"/>
>   <action application="sleep" data="1000"/>
>   <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>

and

<action application="voicemail" data="default ${domain_name} $1"/>

>   </condition>
>     </extension>

HTH
--FC



From dave at 3c.co.uk  Mon Mar  8 10:46:22 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 15:46:22 -0300
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
Message-ID: <AF01ECDC33A74538AEB69214232F3E51@DELL9>

Hi Aloysius,

Best guess is that
<action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
isn't what you meant - ${dialed_extension} will be 9920x, not 20x.

You can fix this by changing the regex to
<condition field="destination_number" expression="^\*99(20[0-9])$">
(the bit in ()s will end up in $1) and the line above to
<action application="voicemail" data="default ${domain_name} $1"/>

Cheers --

Dave
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 3:23 PM
  Subject: [Freeswitch-users] Send a call directly to voicemail Transfer Help


  Hi All, 


  I would like to setup a dial plan, Dial *99[extension number]  send a call directly to voicemail.


   I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed.


      <extension name="voicemail_transfer">
        <condition field="destination_number" expression="^(\*9920[0-9])$">
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
   </condition>
      </extension>
  But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209


  Please let me know where the mistake is.


  Thanks
  Lloyd


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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From grevenx at me.com  Mon Mar  8 10:52:57 2010
From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Mon, 08 Mar 2010 19:52:57 +0100
Subject: [Freeswitch-users] FreeSWITCH cluster with IM/Presence
Message-ID: <9FD45932-DD45-4370-A08B-9C06C6D3289C@me.com>

Hi.

I currently have a project where I'm researching how to establish a clustered and scalable platform with:
- registration
- presence/blf between eyebeam/bria and Linksys SPA962 clients
- IM between eyebeam/bria clients.
- calling between the clients

The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". 
How the registrations and calls etc. are balanced between them is not decided
yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here?
The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers
(which can be scaled by adding n-nodes).

I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, 
and what it would take to make something like this work. If there's any part of FS that is lacking some implementation
to make such a scenario work, we are interested in providing the funding to support those features.
I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would
be helpful in this scenario or not.

This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers 
to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to 
be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to.

Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence.
I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform.


Best regards,
Even Andr? Fiskvik
CTO - Oyatel AS




From grevenx at me.com  Mon Mar  8 11:11:42 2010
From: grevenx at me.com (=?windows-1252?Q?Even_Andr=E9_Fiskvik?=)
Date: Mon, 08 Mar 2010 20:11:42 +0100
Subject: [Freeswitch-users] server to server jingle
In-Reply-To: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
References: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
Message-ID: <147756F3-FD28-45C3-BF2E-DF1DA6399CAC@me.com>

Perhaps you should write a bit about what you're exactly trying to achieve and
if it doesn't seem to "work as expected", describe what issues you are having?

Best regards,
Even Andr? Fiskvik

On 2. mars 2010, at 05.38, Madovsky wrote:

> Hi all,
>  
> is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
> I read this article
> http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
> but no example
> and this one
> http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
> but seems to doesn't work as expected.
>  
> Thanks !
>  
> Franck
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Mon Mar  8 11:26:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 11:26:06 -0800
Subject: [Freeswitch-users] Conference Recording
In-Reply-To: <4B948B0E.8050909@todandlorna.com>
References: <f8d0fbe51003071628i55ac8118j96e36fa31c96a499@mail.gmail.com>
	<4B948B0E.8050909@todandlorna.com>
Message-ID: <87f2f3b91003081126q591662eq476604ddc5c51afd@mail.gmail.com>

Sorry, I still need to x-code and upload Tod's wonderful discussion of
mod_fifo. :)
-MC

On Sun, Mar 7, 2010 at 9:28 PM, Tod Hansmann <freeswitch.org at todandlorna.com
> wrote:

>  For the whole thing?  I know parts were recorded by different people.  I'm
> not sure about the whole thing.
>
> -Tod Hansmann
>
>
> On 3/7/2010 5:28 PM, Josh Rivers wrote:
>
> On last week's conference call, someone was recording, did that recording
> get put online somewhere?
>
>  Josh
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Mar  8 11:27:14 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 14:27:14 -0500
Subject: [Freeswitch-users] server to server jingle
References: <7DB2DF3BDD8F4752A710A08EF9EE4B33@MOBILEE1705>
	<147756F3-FD28-45C3-BF2E-DF1DA6399CAC@me.com>
Message-ID: <2161970E983E4410A1695D4D4E64430E@MOBILEE1705>

I wanted only to avoid to write an xml for every client who uses xmpp.
nevermind, I'm too novice on Openfire and don't want to use anymore time to learn it
since I need to learn more FS.

Thanks for your answer

Franck
  ----- Original Message ----- 
  From: Even Andr? Fiskvik 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 2:11 PM
  Subject: Re: [Freeswitch-users] server to server jingle


  Perhaps you should write a bit about what you're exactly trying to achieve and
  if it doesn't seem to "work as expected", describe what issues you are having?


  Best regards,
  Even Andr? Fiskvik


  On 2. mars 2010, at 05.38, Madovsky wrote:


    Hi all,

    is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible)
    I read this article
    http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working
    but no example
    and this one
    http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&Itemid=2
    but seems to doesn't work as expected.

    Thanks !

    Franck

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





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


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From lloyd.aloysius at gmail.com  Mon Mar  8 11:28:37 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 14:28:37 -0500
Subject: [Freeswitch-users] Send a call directly to voicemail Transfer
	Help
In-Reply-To: <AF01ECDC33A74538AEB69214232F3E51@DELL9>
References: <8a19bf2e1003081023t3e786df4m4fa8a2858eac7ad@mail.gmail.com>
	<AF01ECDC33A74538AEB69214232F3E51@DELL9>
Message-ID: <8a19bf2e1003081128v55a993a4m42d6860bf6657162@mail.gmail.com>

Thank you for all the suggestions.

David & Frank suggestion solve the problem. Here is the working dial plan

  <extension name="voicemail_transfer">
<condition field="destination_number" expression="^\*99(20[0-9])$">
          <action application="answer"/>
          <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain_name} $1"/>
         </condition>
         </extension>


Thank you again.

Regards.
Lloyd


On Mon, Mar 8, 2010 at 1:46 PM, David Knell <dave at 3c.co.uk> wrote:

>  Hi Aloysius,
>
> Best guess is that
> <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
> isn't what you meant - ${dialed_extension} will be 9920x, not 20x.
>
> You can fix this by changing the regex to
> <condition field="destination_number" expression="^\*99(20[0-9])$">
> (the bit in ()s will end up in $1) and the line above to
> <action application="voicemail" data="default ${domain_name} $1"/>
>
> Cheers --
>
> Dave
>
> ----- Original Message -----
> *From:* Aloysius Lloyd <lloyd.aloysius at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 08, 2010 3:23 PM
> *Subject:* [Freeswitch-users] Send a call directly to voicemail Transfer
> Help
>
> Hi All,
>
> I would like to setup a dial plan, Dial *99[extension number]  send a call
> directly to voicemail.
>
>  I setup the following Dial Plan . But when I transfer the call
> *99[extension number] the call failed.
>
>      <extension name="voicemail_transfer">
>       <condition field="destination_number" expression="^(\*9920[0-9])$">
>  <action application="answer"/>
>  <action application="sleep" data="1000"/>
>  <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>  </condition>
>     </extension>
>  But when I transfer the call *99[extension number] the call failed.My
> extensions range 201 - 209
>
> Please let me know where the mistake is.
>
> Thanks
> Lloyd
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Mon Mar  8 11:42:12 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 19:42:12 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion on
	Polycom IP6000 handsets
Message-ID: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>

This is almost certainly more of a Polycom issue than a FreeSWITCH one
but I'll mention it here on the offchance that someone else has
experienced the following:

Loud G.711a calls that are being transcoded to G.722 are massively
distorted on Polycom IP6000 handsets and on all other handsets the
call just sounds loud and isn't distorted. This is most common in
conference calls where callers are using a mixture of codecs.

If I disable G.722 the Polycom sounds fine even though the G.711a call
is still too loud and if I lower the volume of the conference the
distortion still happens.

So, has anyone else experienced this and if so, did you find a solution?

Regards,
Dan Lane



From m.sobkow at marketelsystems.com  Mon Mar  8 11:46:09 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 08 Mar 2010 13:46:09 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
Message-ID: <4B955401.1030004@marketelsystems.com>

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From brian at freeswitch.org  Mon Mar  8 11:51:39 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 13:51:39 -0600
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
Message-ID: <90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>

I'm going to guess you're using 1.0.4?

/b

On Mar 8, 2010, at 1:42 PM, Dan Lane wrote:

> This is almost certainly more of a Polycom issue than a FreeSWITCH one
> but I'll mention it here on the offchance that someone else has
> experienced the following:
> 
> Loud G.711a calls that are being transcoded to G.722 are massively
> distorted on Polycom IP6000 handsets and on all other handsets the
> call just sounds loud and isn't distorted. This is most common in
> conference calls where callers are using a mixture of codecs.
> 
> If I disable G.722 the Polycom sounds fine even though the G.711a call
> is still too loud and if I lower the volume of the conference the
> distortion still happens.
> 
> So, has anyone else experienced this and if so, did you find a solution?
> 
> Regards,
> Dan Lane
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From pjintheusa at gmail.com  Mon Mar  8 12:04:26 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 8 Mar 2010 15:04:26 -0500
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
Message-ID: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>

You could use the api:

 FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
string apiResult = string.Empty;
string Uuid;
string NumberToDial = "3475558308";
string OutgoingCallerID = "2155556666";

Uuid = fsApi.ExecuteString("create_uuid");

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
Uuid, OutgoingCallerID, NumberToDial));






On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:

> Hi,
> I am trying to create a mod_managed API application that takes 2 phone
> numbers as params.  These numbers should then be used to make 2 calls and to
> bridge them. The issue i am having is working out how to place the first
> call leg from in mod_managed.  All the methods i have found are for the
> second leg and require you to pass in the first call leg.
>
> I know there are easier way to do this out of the managed code but i want
> it in the managed code as this is just a proof of concept to prove we can do
> a callback like system via managed code to be hooked into other external
> systems.
> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>
> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
> while (s.ready()) {
> // The call has been answered
> }
>
> This seems to be exactly what i am looking for but seems to be missing from
> mod_managed. I am hoping someone can tell me how to do something similar in
> mod_managed since even if it is not part of the managed code mod_managed is
> meant to have the native access as well so i would assume it would be
> possible.
>
> Thanks in advance
>
> Chris
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From matt at packetfoundry.net  Mon Mar  8 09:27:42 2010
From: matt at packetfoundry.net (Matt Putnam)
Date: Mon, 08 Mar 2010 11:27:42 -0600
Subject: [Freeswitch-users] mod_spidermonkey
Message-ID: <4B95338E.8030709@packetfoundry.net>

http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel

Has anyone found a solution to the above issue? I am having the same 
issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.


Thanks,
Matt



From brian at freeswitch.org  Mon Mar  8 12:33:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 14:33:52 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B95338E.8030709@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
Message-ID: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>

cd /usr/local/freeswitch/mod
ldd mod_spidermonkey.so

And post it to that jira.

/b

On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:

> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
> 
> Has anyone found a solution to the above issue? I am having the same 
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
> 
> 
> Thanks,
> Matt

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From chris.chen2004 at gmail.com  Mon Mar  8 12:39:19 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 15:39:19 -0500
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B95338E.8030709@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
Message-ID: <507898381003081239m4196f0ccn81dc7bbc2f8874f6@mail.gmail.com>

I am having the same problem as reported in the JIRA, although this has been
about a week. Still the same on latest SVN trunk.
I am running Centos 5.4 X86_64.
Thanks,

Chris

On Mon, Mar 8, 2010 at 12:27 PM, Matt Putnam <matt at packetfoundry.net> wrote:

>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chris.chen2004 at gmail.com  Mon Mar  8 12:43:19 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 8 Mar 2010 15:43:19 -0500
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
Message-ID: <507898381003081243j2d73456ei4b04fda5c8fbfef3@mail.gmail.com>

Hi Brian, I added the information from my Centos 5.4 to the JIRA, please see
the comment.
Thanks,
Chris

On Mon, Mar 8, 2010 at 3:33 PM, Brian West <brian at freeswitch.org> wrote:

> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
> And post it to that jira.
>
> /b
>
> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From null at invalid.name  Mon Mar  8 12:46:30 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 20:46:30 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
Message-ID: <b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>

On Mon, Mar 8, 2010 at 7:51 PM, Brian West <brian at freeswitch.org> wrote:
> I'm going to guess you're using 1.0.4?

I am, was this fixed in 1.0.5?

While I have 1.0.5 boxes (which identify themselves as 1.0.trunk)
coming out of my ears the Polycoms are all overseas and in-use so
getting them temporarily connected to a 1.0.5 box is troublesome.



From paul.gore.j at gmail.com  Mon Mar  8 12:48:56 2010
From: paul.gore.j at gmail.com (paul gore)
Date: Mon, 8 Mar 2010 15:48:56 -0500
Subject: [Freeswitch-users] XML curl and multiple extension in dialplan
In-Reply-To: <SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>
References: <da4ab26e1003071103j2f2471bcxa8874eec7f09fecb@mail.gmail.com>
	<da4ab26e1003071413u5c0f2620iad430ca7c2661026@mail.gmail.com>
	<5632B84B-4513-4C02-BABC-CC1C0C48A078@freeswitch.org>
	<da4ab26e1003071730o51cc1b9t4241508f3e40f45a@mail.gmail.com>
	<84088BC7-954F-4830-959B-989FDD062612@freeswitch.org>
	<da4ab26e1003071803j561e892bnb6a3677f97742458@mail.gmail.com>
	<A8C4E4EA-0798-47DC-8A14-FCDAC6CD2698@freeswitch.org>
	<SNT124-W11FDF941BE500154CBD827BB350@phx.gbl>
	<da4ab26e1003080922v7dc3f2c6r18e4f27a3cec4b1e@mail.gmail.com>
	<SNT124-W494B3D07060DA83BA4AC72BB350@phx.gbl>
Message-ID: <da4ab26e1003081248l37fa5716t72d36a897747a5a5@mail.gmail.com>

Thanks a lot Paulo!
It does work, we have updated to the latest SVN and checked cs_routing and
apparently discovered a problem with response generating script, it wasn't
updated after recent DB change.


On Mon, Mar 8, 2010 at 1:31 PM, Paulo Vicentini <vizentini at hotmail.com>wrote:

>  Hello,
>
> FreeSWITCH version: 1.0.trunk (16870M)
>
> I have replied FS even with extensions you have emailed here (in my case I
> have data="sofia/gateway/myGW/888 at conference.freeswitch.org/888") - it
> works
>
> You may need to give a look at "CS_ROUTING" state.
>
>
> best,
> Paulo
> ------------------------------
> Date: Mon, 8 Mar 2010 12:22:37 -0500
> From: paul.gore.j at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] XML curl and multiple extension in dialplan
>
> It's in the Wiki: http://wiki.freeswitch.org/wiki/Mod_xml_curl - "You will
> only need to send one extension back..."
> In my real dial plan I do use unique names for the extensions, but as I
> mentioned FS only executes first extension from the response. If I place
> same extensions in a static XML they work as  expected.
> I use FreeSWITCH Version 1.0.trunk (16573M). Which one do you use? Can you
> post a sample response you generate, please?
>
>
> On Mon, Mar 8, 2010 at 10:42 AM, Paulo Vicentini <vizentini at hotmail.com>wrote:
>
>
> Hi
>
> "As per FS wiki we can only return one XML extension upon FS request. "
>
> Could you please tell me where did you find that info? I do return several
> extensions by request and FS works as expect (Am I wrong doing that?)
>
> "Now we need to return a set of extensions with same number condition and
> context but with different second condition so that FS can pick up the right
> one.
> Is it possible at all? And if yes how response XML with multiple extensions
> should look like?"
>
> Yes, It seems to be possible (Am I wrong?) but notice that you are giving
> the  same extension name for all your extensions:
>
>  <extension name="test9">
>
>
> I think this can mislead dialplan_hunt in mod_dialplan_xml...
> Give each extension a unique name, and FS will parse/regex all them for
> you...
>
>
> All the best,
> Paulo
>
> > From: brian at freeswitch.org
> > Date: Sun, 7 Mar 2010 20:17:19 -0600
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] XML curl and multiple extension in
> dialplan
>
> >
> > Sounds like you zigged where you should have zagged.
> >
> > /b
> >
> > On Mar 7, 2010, at 8:03 PM, paul gore wrote:
> >
> > > :-))
> > > I used it a lot. Unfortunately we do not store did number as a field,
> as I said we store complete extension XML, with extension number and context
> as a key.
> > > So essentially I will retrieve a bunch of extension XMLs which I need
> to parse and run against regex matcher to find out the one, which is what
> exactly FS is doing when XML dial plan is defined in a static file.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From matt at packetfoundry.net  Mon Mar  8 12:49:38 2010
From: matt at packetfoundry.net (Matt Putnam)
Date: Mon, 08 Mar 2010 14:49:38 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
Message-ID: <4B9562E2.7000207@packetfoundry.net>

Posted mine up as well.

On 3/8/2010 2:33 PM, Brian West wrote:
> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
> And post it to that jira.
>
> /b
>
> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>>
>> Has anyone found a solution to the above issue? I am having the same
>> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>>
>>
>> Thanks,
>> Matt
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From brian at freeswitch.org  Mon Mar  8 13:19:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:19:18 -0600
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
	<b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
Message-ID: <C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>

It was a bug in spandsp codecs thats been fixed for sometime.

/b

On Mar 8, 2010, at 2:46 PM, Dan Lane wrote:

> I am, was this fixed in 1.0.5?
> 
> While I have 1.0.5 boxes (which identify themselves as 1.0.trunk)
> coming out of my ears the Polycoms are all overseas and in-use so
> getting them temporarily connected to a 1.0.5 box is troublesome.




From javieraristizabal at gmail.com  Mon Mar  8 13:38:39 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:38:39 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
Message-ID: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>

Hi folks!!
I have an incoming call being bridged to an outgoing call ann i'm trying to
pass the phone number of the incoming caller as the caller id on the leg B.
But it doesn't work.

Here is my dialplan:

<extension name="incoming">
        <condition field="destination_number" expression="^(3055908934)$">
                <action application="info"/>
                <action application="set"
data="effective_callee_id_number=${caller_id_number}"/>
                <action application="bridge"
data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
         </condition>
    </extension>

Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368


This is the INVITE on the bLeg:

INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
Max-Forwards: 68
From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
;transport=udp>;tag=K6U13Hy2rpgBe
To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
>
Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
CSeq: 127923447 INVITE
Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 250
X-FS-Support: update_display
Remote-Party-ID: "Javier"
<sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
s=FreeSWITCH
c=IN IP4 66.231.242.173
t=0 0
m=audio 23516 RTP/AVP 18 101 13
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20


And i'm on FreeSWITCH Version 1.0.trunk (16921)


Many thanks


/Javier
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From null at invalid.name  Mon Mar  8 13:41:53 2010
From: null at invalid.name (Dan Lane)
Date: Mon, 8 Mar 2010 21:41:53 +0000
Subject: [Freeswitch-users] g711 to g722 transcoding causing distortion
	on Polycom IP6000 handsets
In-Reply-To: <C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>
References: <b042b23b1003081142nb18f3c2t6375558748ca5906@mail.gmail.com>
	<90471E30-EBB8-40A6-83F0-16164222EE33@freeswitch.org>
	<b042b23b1003081246y32f7eb91xde2acfc0af5ef8e6@mail.gmail.com>
	<C33A005D-F873-441B-93FE-D2F7844D9B76@freeswitch.org>
Message-ID: <b042b23b1003081341o78b23247jad74aabf00c82a76@mail.gmail.com>

On Mon, Mar 8, 2010 at 9:19 PM, Brian West <brian at freeswitch.org> wrote:
> It was a bug in spandsp codecs thats been fixed for sometime.

Thanks for the quick feedback... I'll try and get that box upgraded ASAP :)



From dule.maillist at gmail.com  Mon Mar  8 13:50:26 2010
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 8 Mar 2010 16:50:26 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
Message-ID: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>

Try using the 'export' app rather than 'set'.

Dan

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Hi folks!!
> I have an incoming call being bridged to an outgoing call ann i'm trying to
> pass the phone number of the incoming caller as the caller id on the leg B.
> But it doesn't work.
>
> Here is my dialplan:
>
> <extension name="incoming">
>         <condition field="destination_number" expression="^(3055908934)$">
>                 <action application="info"/>
>                 <action application="set"
> data="effective_callee_id_number=${caller_id_number}"/>
>                 <action application="bridge"
> data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
>          </condition>
>     </extension>
>
> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
>
>
> This is the INVITE on the bLeg:
>
> INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
>  Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
> Max-Forwards: 68
> From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
> ;transport=udp>;tag=K6U13Hy2rpgBe
> To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
> >
> Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
> CSeq: 127923447 INVITE
> Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 250
> X-FS-Support: update_display
> Remote-Party-ID: "Javier" <sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
> s=FreeSWITCH
> c=IN IP4 66.231.242.173
> t=0 0
> m=audio 23516 RTP/AVP 18 101 13
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
>
> And i'm on FreeSWITCH Version 1.0.trunk (16921)
>
>
> Many thanks
>
>
> /Javier
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 13:52:02 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:52:02 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
Message-ID: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>

chances are your provider wants the caller id in the from field.

/b

On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:

> Hi folks!! 
> I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work.
> 




From Mailings at kh-dev.de  Mon Mar  8 13:54:10 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 22:54:10 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>

Hi

you could also use newest trunk of t38modem + trunk of OPAL.
The newest trunk version(s) use SpanDSP as fax engine and it works more reliable than the previous t38 engine (I think it was integrated in OPAL).
It also supports fallback to t30 audio fax if you're connected to ISDN network.

Together with HylaFAX+ the success rate is ok (unfortunately not perfect).

Klaus

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Monday, March 08, 2010 5:25 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

On Mon, Mar 8, 2010 at 10:57 AM, Steve Underwood <steveu at coppice.org> wrote:
> We already have a better FAX engine than that. What we struggle to find
> is the time to properly integrate it with FS.
>
> Steve

  Thank you, that's what I was expecting.

  Without going into grueling detail it isn't clear why T.38 gateway
functionality (for example) is missing in FreeSWITCH.  I know SpanDSP
is available and most likely has this ability.  I just wasn't sure why
it's missing in FS.  This is also a mere curiosity for me; I don't
have much interest in fax functionality and I don't follow it too
closely.

  Thanks again for the response and explanation.

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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





From javieraristizabal at gmail.com  Mon Mar  8 13:58:46 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:58:46 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
Message-ID: <b6f11f301003081358t4ba2d935l9ce408a7425fa5d1@mail.gmail.com>

HI Dan. Thanks.  I did it with the same result.


/Javier

On Mon, Mar 8, 2010 at 4:50 PM, Dan Le <dule.maillist at gmail.com> wrote:

> Try using the 'export' app rather than 'set'.
>
> Dan
>
> 2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Hi folks!!
>> I have an incoming call being bridged to an outgoing call ann i'm trying
>> to pass the phone number of the incoming caller as the caller id on the leg
>> B. But it doesn't work.
>>
>> Here is my dialplan:
>>
>> <extension name="incoming">
>>         <condition field="destination_number" expression="^(3055908934)$">
>>                 <action application="info"/>
>>                 <action application="set"
>> data="effective_callee_id_number=${caller_id_number}"/>
>>                 <action application="bridge"
>> data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/>
>>          </condition>
>>     </extension>
>>
>> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368
>>
>>
>> This is the INVITE on the bLeg:
>>
>> INVITE sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>SIP/2.0
>>  Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp
>> Max-Forwards: 68
>> From: "Javier" <sip:xcast at wholesale.xcastlabs.com<sip%3Axcast at wholesale.xcastlabs.com>
>> ;transport=udp>;tag=K6U13Hy2rpgBe
>> To: <sip:17862065658 at wholesale.xcastlabs.com<sip%3A17862065658 at wholesale.xcastlabs.com>
>> >
>> Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2
>> CSeq: 127923447 INVITE
>> Contact: <sip:gw+xcast at 66.231.242.173:5080;transport=udp;gw=xcast>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 250
>> X-FS-Support: update_display
>> Remote-Party-ID: "Javier" <sip:3052606062 at wholesale.xcastlabs.com<sip%3A3052606062 at wholesale.xcastlabs.com>
>> >;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173
>> s=FreeSWITCH
>> c=IN IP4 66.231.242.173
>> t=0 0
>> m=audio 23516 RTP/AVP 18 101 13
>> a=rtpmap:18 G729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>>
>> And i'm on FreeSWITCH Version 1.0.trunk (16921)
>>
>>
>> Many thanks
>>
>>
>> /Javier
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Mar  8 13:59:30 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 15:59:30 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<914fc92a1003081350r451e1bf4jd3cd5644c79d4b32@mail.gmail.com>
Message-ID: <BDACCFA5-0901-4B62-80BE-3EAF5FAAEC05@freeswitch.org>

No we by default already pass it... no need to export vs set.. plus effective_caller_id_number/name is set on the A-LEG and auto exported for you to the B-LEG when created.

His idiot provider wants it in the from field I suspect.

/b

On Mar 8, 2010, at 3:50 PM, Dan Le wrote:

> Try using the 'export' app rather than 'set'.
> 
> Dan




From javieraristizabal at gmail.com  Mon Mar  8 13:59:47 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 16:59:47 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
Message-ID: <b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>

Hi Brian. Thanks, yeah it is possible. Is there a way to do that?

/Javier

On Mon, Mar 8, 2010 at 4:52 PM, Brian West <brian at freeswitch.org> wrote:

> chances are your provider wants the caller id in the from field.
>
> /b
>
> On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:
>
> > Hi folks!!
> > I have an incoming call being bridged to an outgoing call ann i'm trying
> to pass the phone number of the incoming caller as the caller id on the leg
> B. But it doesn't work.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Mon Mar  8 13:59:34 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 08:59:34 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com> 
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
Message-ID: <9394d42f1003081359u5cb0f2e0q90695d1abb0be9b4@mail.gmail.com>

Thanks phillip.
I assume i could then use the command new ManagedSession(uuid); to get
access to the session after it is created.  I will give it a go later today
but i think you are right that should work.

For a mater of interest would you have any idea where in the source code for
FS i could find the implementation for the originate method called via the
API command you just suggested?  If the implementation is not too
complicated I may look at adding it to the managed code as the managed code
is meant to have access to the native APIs so it should be able to issue the
same commands i would have thought. If it is impossible i will stick with
the way you just suggested.

Thanks

Chris

On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> You could use the api:
>
>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
> string apiResult = string.Empty;
> string Uuid;
> string NumberToDial = "3475558308";
> string OutgoingCallerID = "2155556666";
>
> Uuid = fsApi.ExecuteString("create_uuid");
>
> apiResult = fsApi.Execute("originate",
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
> Uuid, OutgoingCallerID, NumberToDial));
>
>
>
>
>
>
> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Hi,
>> I am trying to create a mod_managed API application that takes 2 phone
>> numbers as params.  These numbers should then be used to make 2 calls and to
>> bridge them. The issue i am having is working out how to place the first
>> call leg from in mod_managed.  All the methods i have found are for the
>> second leg and require you to pass in the first call leg.
>>
>> I know there are easier way to do this out of the managed code but i want
>> it in the managed code as this is just a proof of concept to prove we can do
>> a callback like system via managed code to be hooked into other external
>> systems.
>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>
>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>> while (s.ready()) {
>> // The call has been answered
>> }
>>
>> This seems to be exactly what i am looking for but seems to be missing
>> from mod_managed. I am hoping someone can tell me how to do something
>> similar in mod_managed since even if it is not part of the managed code
>> mod_managed is meant to have the native access as well so i would assume it
>> would be possible.
>>
>> Thanks in advance
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar  8 14:00:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:00:13 -0800
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
Message-ID: <87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>

On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:

> chances are your provider wants the caller id in the from field.
>
> /b
>
This sounds more likely. Javar tried the export already and that didn't
help. What is the proper procedure for setting the From field? Can you just
export sip_from_user when doing the bridge?
-MC


>
> On Mar 8, 2010, at 3:38 PM, Javier Aristiz?bal wrote:
>
> > Hi folks!!
> > I have an incoming call being bridged to an outgoing call ann i'm trying
> to pass the phone number of the incoming caller as the caller id on the leg
> B. But it doesn't work.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  8 14:02:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:02:26 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
Message-ID: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>

Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).




From brian at freeswitch.org  Mon Mar  8 14:03:49 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:03:49 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<87f2f3b91003081400j7874a025ra8dd5bec89a50e21@mail.gmail.com>
Message-ID: <029A5E59-4C83-451F-91AB-465C4585EBFA@freeswitch.org>

Try sip_from_uri

/b

On Mar 8, 2010, at 4:00 PM, Michael Collins wrote:

> 
> 
> On Mon, Mar 8, 2010 at 1:52 PM, Brian West <brian at freeswitch.org> wrote:
> chances are your provider wants the caller id in the from field.
> 
> /b
> This sounds more likely. Javar tried the export already and that didn't help. What is the proper procedure for setting the From field? Can you just export sip_from_user when doing the bridge?
> -MC

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From brian at freeswitch.org  Mon Mar  8 14:04:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:04:33 -0600
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
Message-ID: <EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>

if all else fails try reading do_invite in sofia_glue.c you can really change just about ANYTHING to do with the invite with variables... be careful some let you totally break the packet where it won't even work.  :P

/b

On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:

> Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
> 
> /Javier




From javieraristizabal at gmail.com  Mon Mar  8 14:12:39 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:12:39 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
Message-ID: <b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>

Thanks Brian, let me try your ideas.

/Javier

On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian at freeswitch.org> wrote:

> if all else fails try reading do_invite in sofia_glue.c you can really
> change just about ANYTHING to do with the invite with variables... be
> careful some let you totally break the packet where it won't even work.  :P
>
> /b
>
> On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:
>
> > Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
> >
> > /Javier
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From will.traenkle at yahoo.com  Mon Mar  8 14:23:45 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 14:23:45 -0800 (PST)
Subject: [Freeswitch-users] auto attendant -> hunt group -> external number
	does not ring
Message-ID: <302537.41702.qm@web57609.mail.re1.yahoo.com>

FS Community,

I created an extension
704
 
I created a
hunt group 7004 that first rings 704 then rings my cell phone 16194548924.
 
I created an
auto attendant 5002 with an option of 704 that goes to hunt group 7004.
 
1)     What does
work: When I call the hunt group directly by dialing 7004 internally, the hunt
group first rings 704 then rings my cell phone successfully.  This is perfect.
 
2)     What does not
work: When I call the main auto attendant from either internally by dialing 5002
or externally by dialing my main number, and dial 704, my extension rings and
after not answering my cell phone does not ring.
 
What I am expecting
in #2 is for my extension 704 to ring and if I do not pick up then my cell
phone will ring just like it does in #1.
 
My knowledge
is limited in this area and if you could point me in the right direction, that
would be great.  I a have tried
everything but with no luck.

Thanks,

-Will



      
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From Mailings at kh-dev.de  Mon Mar  8 14:25:39 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 23:25:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>

Believe it or not...
I heard from people out there who don't have email.
But they use faxing...


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, March 08, 2010 11:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).


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





From javieraristizabal at gmail.com  Mon Mar  8 14:28:53 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:28:53 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
Message-ID: <b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>

Brian. i did with the sip_from_uri, and it is working..

Many thanks

/Javier

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Thanks Brian, let me try your ideas.
>
> /Javier
>
>
> On Mon, Mar 8, 2010 at 5:04 PM, Brian West <brian at freeswitch.org> wrote:
>
>> if all else fails try reading do_invite in sofia_glue.c you can really
>> change just about ANYTHING to do with the invite with variables... be
>> careful some let you totally break the packet where it won't even work.  :P
>>
>> /b
>>
>> On Mar 8, 2010, at 3:59 PM, Javier Aristiz?bal wrote:
>>
>> > Hi Brian. Thanks, yeah it is possible. Is there a way to do that?
>> >
>> > /Javier
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From will.traenkle at yahoo.com  Mon Mar  8 14:31:50 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 14:31:50 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <847524.92329.qm@web57603.mail.re1.yahoo.com>

I am sending this again as I did not see it go through the first time ...


FS Community,


I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

My hunt group code is below:


session.answer();
var domain_name = session.getVariable("domain_name");
var extension = '7000';
var result;
var timeoutpin = 7500;

var objdate = new Date();
var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours
var adjustoperator = "-"; //+ or -
if (adjustoperator == "-") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
if (adjustoperator == "+") {
var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds());
}
var Hours = objdate2.getHours();
var Mins = objdate2.getMinutes();
var Seconds = objdate2.getSeconds();
var Month = objdate2.getMonth() + 1;
var Date = objdate2.getDate();
var Year = objdate2.getYear()
var Day = objdate2.getDay()+1;
var exit = false;

 function get_sofia_contact(extension,domain_name, profile){
        if (profile == "auto") {
                        profile = "internal";
                        session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                        sofia_contact = session.getVariable("sofia_contact_"+extension);

                        if (sofia_contact == "error/user_not_registered") {
                                profile = "external";
                                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                                sofia_contact = session.getVariable("sofia_contact_"+extension);
                        }
        }
        else {
                session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}");
                sofia_contact = session.getVariable("sofia_contact_"+extension);
        }
        console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" );
        return sofia_contact;
 }


 function mycb( session, type, obj, arg ) {
    try {
        if ( type == "dtmf" ) {
          console_log( "info", "digit: "+obj.digit+"\n" );
          if ( obj.digit == "#" ) {
            //console_log( "info", "detected pound sign.\n" );
            exit = true;
            return( false );
          }

          dtmf.digits += obj.digit;

          if ( dtmf.digits.length >= digitmaxlength ) {
            exit = true;
            return( false );
          }
        }
    } catch (e) {
        console_log( "err", e+"\n" );
    }
    return( true );
 } //end function mycb


dialed_extension = session.getVariable("dialed_extension");
domain_name = session.getVariable("domain_name");
domain = session.getVariable("domain");
us_ring = session.getVariable("us-ring");
caller_id_name = session.getVariable("caller_id_name");
caller_id_number = session.getVariable("caller_id_number");
effective_caller_id_name = session.getVariable("effective_caller_id_name");
effective_caller_id_number = session.getVariable("effective_caller_id_number");
outbound_caller_id_name = session.getVariable("outbound_caller_id_name");
outbound_caller_id_number = session.getVariable("outbound_caller_id_number");
session.execute("set", "ringback=${hold_music}");          //set to ringtone
session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone
session.execute("set", "call_timeout=30");
session.execute("export", "call_timeout=30");
session.execute("set", "continue_on_fail=true");
session.execute("set", "hangup_after_bridge=true");

//console_log( "info", "dialed extension:"+dialed_extension+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "us_ring: "+us_ring+".\n" );
//console_log( "info", "domain: "+domain+".\n" );
//console_log( "info", "domain_name: "+domain_name+".\n" );

//console_log( "info", "action call now don't wait for dtmf\n" );
if ( session.ready() ) {
        session.answer();

        sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto");

        session.execute("bridge", sofia_contact_1000);
        session.execute("bridge", "sofia/gateway/flowroute.com/16194548924");

        //timeout
        if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){
                session.execute("info", "");
                session.execute("transfer", "*991000");
        }

        //clear variables
        dialed_extension = "";
        new_extension = "";
        domain_name = "";
        domain = "";
} //end if session.ready


Thanks.

-Will


      
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From lloyd.aloysius at gmail.com  Mon Mar  8 14:33:33 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 17:33:33 -0500
Subject: [Freeswitch-users] What is the best way implement the follow me for
	a extension
Message-ID: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>

Hi All,

I would like to setup follow me dialplan for individual extensions. Some
extensions need a follow-me and some not. If the the follow-me not reach any
live person the call should go to the FreeSWTICH voicemail.I have the
extensions range from 201 -209. Also I am using the following Default dial
plan to call between extensions.

<extension name="Local_Extension">
      <condition field="destination_number" expression="^(20[0-9])$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<action application="bind_meta_app" data="1 b s execute_extension::dx XML
features"/>
<action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML
features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=${network_addr}"/>
-->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
<action application="set" data="continue_on_fail=true"/>
<action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
<!--<action application="export"
data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
var sip_secure_media)}"/>-->
<action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@
${domain_name},${followme}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
      </condition>
 </extension>

I use the wiki samples to make a follow me dial plan for extension 201. Here
is the dial plan.

<extension name="201_followme">
  <condition field="destination_number" expression="201">
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="call_timeout=10"/>
   <action application="bridge" data="sofia/$${domain}/201"/>
   <action application="set" data="call_timeout=13"/>
   <action application="bridge" data="sofia/gateway/voicepulse/14161235555"
/>
   <action application="answer"/>
   <action application="sleep" data="1000"/>
   <action application="voicemail" data="default $${domain} 201"/>
  </condition>
</extension>


Call to extension 201 , there are two dialplan matching the request.
Either Local_Extension or follwme_201. Which order freeswtich select?

Thank you for your help.

Thanks,
Lloyd
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From msc at freeswitch.org  Mon Mar  8 14:34:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:34:43 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
Message-ID: <87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:25 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

> Believe it or not...
> I heard from people out there who don't have email.
> But they use faxing...
>
Whenever you find people like that, ask them if they've ever heard of the
Internet. I hear this Internet thing has a bright future.
-MC
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From brian at freeswitch.org  Mon Mar  8 14:35:19 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:35:19 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
Message-ID: <9F469ABB-1C87-4D11-85D3-FC2B35A1F516@freeswitch.org>

Lets get a search party and go find them and bust that rock they are hiding under.  :P

/b

On Mar 8, 2010, at 4:25 PM, Klaus Hochlehnert wrote:

> Believe it or not...
> I heard from people out there who don't have email.
> But they use faxing...




From msc at freeswitch.org  Mon Mar  8 14:35:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 14:35:39 -0800
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
	<b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
Message-ID: <87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>

2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Brian. i did with the sip_from_uri, and it is working..
>
> Many thanks
>
> /Javier
>

FYI,

I couldn't find this specific tip in the wiki so I added it here:
http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri

-MC
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From brian at freeswitch.org  Mon Mar  8 14:39:08 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:39:08 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA48@srv02.khdev.corp>
	<87f2f3b91003081434r1e351185ma29ce8b94d99aa56@mail.gmail.com>
Message-ID: <78F1F8A9-6A65-4F03-BD72-D948CC241781@freeswitch.org>

Don't know about that my parents always told me that computers were a waste of time and the internet was never gonna be a big deal.

So I'm guessing I'm lucky I didn't listen to my parents.

/b

On Mar 8, 2010, at 4:34 PM, Michael Collins wrote:

> Whenever you find people like that, ask them if they've ever heard of the Internet. I hear this Internet thing has a bright future. 
> -MC




From gkuri at ieee.org  Mon Mar  8 14:39:29 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 8 Mar 2010 14:39:29 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>

While I'd love to see FAXing just die, especially with the amount of
headache associated with it, the reality is that specific organizations
still rely on it heavily on a daily basis (ie Health Care and
Banking/Finance Industry). Plus they can legally get away with FAXing
personal data directly between machines, since the FAX transmission doesn't
seem to be directly covered by all their respective privacy regultions (ie
HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
of these organizations just don't have the internal expertise to setup
end-to-end encrypted email nor the impetus to deal with it, since FAXing
works fine in their eyes. I think it's going to be around quite a while
longer unless they actually stop making the FAX machines and force people to
use something else.

Cheers,
Gabe

On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

> Thats unacceptable!!!  Wait its faxing... can't we just let it DIE
> already??  They have this thing called email... you should check it out...
> works great... even lets you attach video, audio and color images.... oh and
> text... dreary old text.... but hey you can mark it all up in HTML and fancy
> fonts now and make it anoying^H^H^H^H^H^H^Hpretty.
>
> /b
>
> On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:
>
> > Together with HylaFAX+ the success rate is ok (unfortunately not
> perfect).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar  8 14:46:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 16:46:44 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>

There is no such thing as security when humans are involved in the relay of information.  Thats fact.

HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are involved its not secure.

/b

On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:

> While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.
> 
> Cheers,
> Gabe




From javieraristizabal at gmail.com  Mon Mar  8 14:47:02 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 8 Mar 2010 17:47:02 -0500
Subject: [Freeswitch-users] CID Question: How pass the CID?
In-Reply-To: <87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>
References: <b6f11f301003081338g408d9d18red3a6ff4677a7eb3@mail.gmail.com>
	<DC7A7C77-A9D7-47FB-B830-FCD4EC8BE04B@freeswitch.org>
	<b6f11f301003081359l3b4c12ffj99f5dbf637699c22@mail.gmail.com>
	<EAEEB294-B4E5-4E75-B618-0AA7C8245177@freeswitch.org>
	<b6f11f301003081412n5c2f494etce716a283dda6e27@mail.gmail.com>
	<b6f11f301003081428m76525e1ehc9d678033bed93aa@mail.gmail.com>
	<87f2f3b91003081435ie5fb465q951490f66a505824@mail.gmail.com>
Message-ID: <b6f11f301003081447r7bc2ab3dy917265f91cef3fd9@mail.gmail.com>

Many Thanks Michael :)

/Javier

On Mon, Mar 8, 2010 at 5:35 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> 2010/3/8 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Brian. i did with the sip_from_uri, and it is working..
>>
>> Many thanks
>>
>> /Javier
>>
>
> FYI,
>
> I couldn't find this specific tip in the wiki so I added it here:
> http://wiki.freeswitch.org/wiki/Variable_sip_from_uri#sip_from_uri
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Mailings at kh-dev.de  Mon Mar  8 14:58:39 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 8 Mar 2010 23:58:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>

Me too. I'd also like to see it die.

But honestly I know a lot of small and medium business companies that really rely on faxing.
And for some parts it's much easier than using email. Just say if you get a piece of paper from your tax office and want to send it to your tax consultant.
- Put it in the fax and send it
or
- Scan this piece of paper, save it to the disk, find it (which is btw the most challenging part for most people) and send it by email (most SMB companies don't even have a scanner attached to their PC)

So most people prefer the faxing way... J

Klaus


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Kuri
Sent: Monday, March 08, 2010 11:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

Cheers,
Gabe
On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

/b

On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).

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

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From dave at 3c.co.uk  Mon Mar  8 15:01:04 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 20:01:04 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <FDDF0EE896B34D1A80151B6032B1716C@DELL9>

And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

--Dave
  ----- Original Message ----- 
  From: Gabriel Kuri 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 7:39 PM
  Subject: Re: [Freeswitch-users] Attrafax


  While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

  Cheers,
  Gabe


  On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

    Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

    /b


    On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

    > Together with HylaFAX+ the success rate is ok (unfortunately not perfect).



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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From Mailings at kh-dev.de  Mon Mar  8 15:02:38 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 00:02:38 +0100
Subject: [Freeswitch-users] Facebook...
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>

Hi Guys,

just saw that you already have a Facebook entry.

Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook?
I'd really like to see all news on Facebook...

Cheers,
Klaus
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From brian at freeswitch.org  Mon Mar  8 15:05:56 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 17:05:56 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <FDDF0EE896B34D1A80151B6032B1716C@DELL9>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
Message-ID: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>

Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me...

/b

On Mar 8, 2010, at 5:01 PM, David Knell wrote:

> And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.
>  
> Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.
>  
> --Dave
> 

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From brian at microcomaustralia.com.au  Mon Mar  8 15:08:56 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Tue, 9 Mar 2010 10:08:56 +1100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
Message-ID: <3c5cf5261003081508p710c2461g245a688d01f55897@mail.gmail.com>

On 9 March 2010 09:39, Gabriel Kuri <gkuri at ieee.org> wrote:
> Plus they can legally get away with FAXing personal data directly between machines [...]

It is not that uncommon to accidentally send a confidential fax to the
wrong telephone number ... :-(

I have a vague recollection of an incident here where it had been
discovered the government in power had been sending confidential faxes
to a member of the opposition for months until they realized their
records hadn't been updated properly since the last election.

Oops.
-- 
Brian May <brian at microcomaustralia.com.au>



From msc at freeswitch.org  Mon Mar  8 15:10:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 15:10:59 -0800
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
Message-ID: <87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I would like to setup follow me dialplan for individual extensions. Some
> extensions need a follow-me and some not. If the the follow-me not reach any
> live person the call should go to the FreeSWTICH voicemail.I have the
> extensions range from 201 -209. Also I am using the following Default dial
> plan to call between extensions.
>
> <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(20[0-9])$">
> <action application="set" data="dialed_extension=$1"/>
>  <action application="export" data="dialed_extension=$1"/>
> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML
> features"/>
> <action application="bind_meta_app" data="2 b s
> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>  <action application="bind_meta_app" data="3 b s execute_extension::cf XML
> features"/>
> <action application="set" data="ringback=${us-ring}"/>
>  <action application="set" data="transfer_ringback=$${hold_music}"/>
> <action application="set" data="call_timeout=30"/>
>  <!-- <action application="set"
> data="sip_exclude_contact=${network_addr}"/> -->
> <action application="set" data="hangup_after_bridge=true"/>
>  <!--<action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
> -->
>  <action application="set" data="continue_on_fail=true"/>
> <action application="hash"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>  <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
>  <!--<action application="export"
> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
> var sip_secure_media)}"/>-->
>  <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
> <action application="bridge" data="user/${dialed_extension}@
> ${domain_name},${followme}"/>
>  <action application="answer"/>
> <action application="sleep" data="1000"/>
>  <action application="voicemail" data="default ${domain_name}
> ${dialed_extension}"/>
>       </condition>
>  </extension>
>
> I use the wiki samples to make a follow me dial plan for extension 201.
> Here is the dial plan.
>
> <extension name="201_followme">
>   <condition field="destination_number" expression="201">
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail=true"/>
>    <action application="set" data="ignore_early_media=true"/>
>    <action application="set" data="call_timeout=10"/>
>    <action application="bridge" data="sofia/$${domain}/201"/>
>    <action application="set" data="call_timeout=13"/>
>    <action application="bridge" data="sofia/gateway/voicepulse/14161235555"
> />
>    <action application="answer"/>
>    <action application="sleep" data="1000"/>
>    <action application="voicemail" data="default $${domain} 201"/>
>   </condition>
> </extension>
>
>
> Call to extension 201 , there are two dialplan matching the request.
> Either Local_Extension or follwme_201. Which order freeswtich select?
>

It will match whichever one comes first in the file. You could, for example,
put this 201 extension above the Local_Extension. You could also put a
custom chan var in the user directory for those who need the follow-me. Then
you could have two conditions, both of which need to match:

<extension name="Local, follow me">
  <condition field="${followme}" expression="^yes$"/>
  <condition field="destination_number expression="^(20\d)$">
    <!-- followme stuff here -->
    ...
  </condition>
</extension>

<extension name="Local No follow me">
  <condition field="${followme}" expression="^$"/>
  <condition field="destination_number expression="^(20\d)$">
    <!-- normal local extension stuff here -->
    ...
  </condition>
</extension>

Then in the user setup in the directory add the followme variable only to
those who need it. You can also add a variable for the followme phone number
if you want. Inside the <variables> node just add whatever you want:

  <variable name="followme" value="yes"/>
  <variable name="followme_number" value="14161235555"/>
Or even
  <variable name="followme_dialstring"
value="sofia/gateway/voicepulse/14161235555"/>

Then you could use ${followme_number} in your bridge like this:
  <action application="bridge" data="sofia/gateway/
voicepulse/${followme_number}" />

Or ${followme_dialstring} like this:
  <action application="bridge" data="${followme_dialstring}" />

Make sense? :)

Give it a try and report back your success/failure/questions/comments/etc.
-MC
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From anthony.minessale at gmail.com  Mon Mar  8 15:20:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Mar 2010 17:20:04 -0600
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B9562E2.7000207@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
	<4B9562E2.7000207@packetfoundry.net>
Message-ID: <191c3a031003081520y1c2c8bf2r9be0121b28273f0e@mail.gmail.com>

Posting bugs to jira then posting to the mailing list asking us to look at
it is bad etiquette btw.
We already have notification when the bug is posted.


On Mon, Mar 8, 2010 at 2:49 PM, Matt Putnam <matt at packetfoundry.net> wrote:

>  Posted mine up as well.
>
>
> On 3/8/2010 2:33 PM, Brian West wrote:
>
> cd /usr/local/freeswitch/mod
> ldd mod_spidermonkey.so
>
>  And post it to that jira.
>
>  /b
>
>  On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>
>
> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>
> Has anyone found a solution to the above issue? I am having the same
> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>
>
> Thanks,
> Matt
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From dave at 3c.co.uk  Mon Mar  8 15:21:53 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 20:21:53 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org><8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
Message-ID: <BDEDAB2A5DB04B96B8BD4C3E78DE128E@DELL9>

Quite.  Douglas Adams hit many nails on the head: here's one of them: "We are stuck with technology when what we really want is just stuff that works."

If you want a real challenge, try to fax a document using a WiFi scanner, a Linux box and Pamfax (or pretty much any other online fax service): by the time you get to xsane, you will be..

--Dave

  Me too. I'd also like to see it die.

   

  But honestly I know a lot of small and medium business companies that really rely on faxing.

  And for some parts it's much easier than using email. Just say if you get a piece of paper from your tax office and want to send it to your tax consultant.

  - Put it in the fax and send it

  or

  - Scan this piece of paper, save it to the disk, find it (which is btw the most challenging part for most people) and send it by email (most SMB companies don't even have a scanner attached to their PC)

   

  So most people prefer the faxing way... J

   

  Klaus

   

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Kuri
  Sent: Monday, March 08, 2010 11:39 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Attrafax

   

  While I'd love to see FAXing just die, especially with the amount of headache associated with it, the reality is that specific organizations still rely on it heavily on a daily basis (ie Health Care and Banking/Finance Industry). Plus they can legally get away with FAXing personal data directly between machines, since the FAX transmission doesn't seem to be directly covered by all their respective privacy regultions (ie HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most of these organizations just don't have the internal expertise to setup end-to-end encrypted email nor the impetus to deal with it, since FAXing works fine in their eyes. I think it's going to be around quite a while longer unless they actually stop making the FAX machines and force people to use something else.

  Cheers,
  Gabe

  On Mon, Mar 8, 2010 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

  Thats unacceptable!!!  Wait its faxing... can't we just let it DIE already??  They have this thing called email... you should check it out... works great... even lets you attach video, audio and color images.... oh and text... dreary old text.... but hey you can mark it all up in HTML and fancy fonts now and make it anoying^H^H^H^H^H^H^Hpretty.

  /b


  On Mar 8, 2010, at 3:54 PM, Klaus Hochlehnert wrote:

  > Together with HylaFAX+ the success rate is ok (unfortunately not perfect).



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

   



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From gavin.henry at gmail.com  Mon Mar  8 15:22:58 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 8 Mar 2010 23:22:58 +0000
Subject: [Freeswitch-users] mod_spidermonkey
In-Reply-To: <4B9562E2.7000207@packetfoundry.net>
References: <4B95338E.8030709@packetfoundry.net>
	<84B95AC3-3312-4555-B1DD-F52881262F9B@freeswitch.org>
	<4B9562E2.7000207@packetfoundry.net>
Message-ID: <13ca621c1003081522h422e225brcda68f4b1ecdc76c@mail.gmail.com>

Hi,

That was my bug report. I gave up asking in the #freeswitch channel
and tried to fix it myself. I'm running the debian build from trunk
that you build yourself to produce debs. Mod_spidermoney compiled
correctly and is now running. I forgot to add this info to the JIRA
ticket as that is the best place for code support/issued rather than
irc.

Thanks,

Gavin.

On 08/03/2010, Matt Putnam <matt at packetfoundry.net> wrote:
> Posted mine up as well.
>
> On 3/8/2010 2:33 PM, Brian West wrote:
>> cd /usr/local/freeswitch/mod
>> ldd mod_spidermonkey.so
>>
>> And post it to that jira.
>>
>> /b
>>
>> On Mar 8, 2010, at 11:27 AM, Matt Putnam wrote:
>>
>>> http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
>>>
>>> Has anyone found a solution to the above issue? I am having the same
>>> issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400.
>>>
>>>
>>> Thanks,
>>> Matt
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From lloyd.aloysius at gmail.com  Mon Mar  8 15:24:16 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 8 Mar 2010 18:24:16 -0500
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
Message-ID: <8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>

Michael,

Thank you for the suggestion. This suggestion make sense and a proper way to
implement the follow me. .

Thanks
Lloyd

On Mon, Mar 8, 2010 at 6:10 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> I would like to setup follow me dialplan for individual extensions. Some
>> extensions need a follow-me and some not. If the the follow-me not reach any
>> live person the call should go to the FreeSWTICH voicemail.I have the
>> extensions range from 201 -209. Also I am using the following Default dial
>> plan to call between extensions.
>>
>> <extension name="Local_Extension">
>>       <condition field="destination_number" expression="^(20[0-9])$">
>> <action application="set" data="dialed_extension=$1"/>
>>  <action application="export" data="dialed_extension=$1"/>
>> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>>  <action application="bind_meta_app" data="1 b s execute_extension::dx
>> XML features"/>
>> <action application="bind_meta_app" data="2 b s
>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>  <action application="bind_meta_app" data="3 b s execute_extension::cf
>> XML features"/>
>> <action application="set" data="ringback=${us-ring}"/>
>>  <action application="set" data="transfer_ringback=$${hold_music}"/>
>> <action application="set" data="call_timeout=30"/>
>>  <!-- <action application="set"
>> data="sip_exclude_contact=${network_addr}"/> -->
>> <action application="set" data="hangup_after_bridge=true"/>
>>  <!--<action application="set"
>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
>> -->
>>  <action application="set" data="continue_on_fail=true"/>
>> <action application="hash"
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>  <action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>> <action application="set"
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>>  <!--<action application="export"
>> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
>> var sip_secure_media)}"/>-->
>>  <action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>> <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name},${followme}"/>
>>  <action application="answer"/>
>> <action application="sleep" data="1000"/>
>>  <action application="voicemail" data="default ${domain_name}
>> ${dialed_extension}"/>
>>       </condition>
>>  </extension>
>>
>> I use the wiki samples to make a follow me dial plan for extension 201.
>> Here is the dial plan.
>>
>> <extension name="201_followme">
>>   <condition field="destination_number" expression="201">
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="set" data="continue_on_fail=true"/>
>>    <action application="set" data="ignore_early_media=true"/>
>>    <action application="set" data="call_timeout=10"/>
>>    <action application="bridge" data="sofia/$${domain}/201"/>
>>    <action application="set" data="call_timeout=13"/>
>>    <action application="bridge"
>> data="sofia/gateway/voicepulse/14161235555" />
>>    <action application="answer"/>
>>    <action application="sleep" data="1000"/>
>>    <action application="voicemail" data="default $${domain} 201"/>
>>   </condition>
>> </extension>
>>
>>
>> Call to extension 201 , there are two dialplan matching the request.
>> Either Local_Extension or follwme_201. Which order freeswtich select?
>>
>
> It will match whichever one comes first in the file. You could, for
> example, put this 201 extension above the Local_Extension. You could also
> put a custom chan var in the user directory for those who need the
> follow-me. Then you could have two conditions, both of which need to match:
>
> <extension name="Local, follow me">
>   <condition field="${followme}" expression="^yes$"/>
>   <condition field="destination_number expression="^(20\d)$">
>     <!-- followme stuff here -->
>     ...
>   </condition>
> </extension>
>
> <extension name="Local No follow me">
>   <condition field="${followme}" expression="^$"/>
>   <condition field="destination_number expression="^(20\d)$">
>     <!-- normal local extension stuff here -->
>     ...
>   </condition>
> </extension>
>
> Then in the user setup in the directory add the followme variable only to
> those who need it. You can also add a variable for the followme phone number
> if you want. Inside the <variables> node just add whatever you want:
>
>   <variable name="followme" value="yes"/>
>   <variable name="followme_number" value="14161235555"/>
> Or even
>   <variable name="followme_dialstring"
> value="sofia/gateway/voicepulse/14161235555"/>
>
> Then you could use ${followme_number} in your bridge like this:
>
>   <action application="bridge" data="sofia/gateway/
> voicepulse/${followme_number}" />
>
> Or ${followme_dialstring} like this:
>   <action application="bridge" data="${followme_dialstring}" />
>
> Make sense? :)
>
> Give it a try and report back your success/failure/questions/comments/etc.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar  8 15:28:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 15:28:50 -0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA49@srv02.khdev.corp>
Message-ID: <87f2f3b91003081528o6d716414h85b607897b2ab13b@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:58 PM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Me too. I?d also like to see it die.
>
>
>
> But honestly I know a lot of small and medium business companies that
> really rely on faxing.
>
> And for some parts it?s much easier than using email. Just say if you get a
> piece of paper from your tax office and want to send it to your tax
> consultant.
>
> - Put it in the fax and send it
>
> or
>
> - Scan this piece of paper, save it to the disk, find it (which is btw the
> most challenging part for most people) and
>
Or get a device that will just do the scan & email in one fell swoop. You
could even encrypt before sending and then the person would have to contact
you to get the key before opening it. Not 100% foolproof but it would
prevent the silly, oops, I sent your personal information to the wrong email
address/fax machine. Just think, you could have the convenience of a fax and
relatively secure storage of incoming and outgoing transmissions. Of course,
wanting to cling to 1970's technology is something I can understand. Who
wouldn't want to keep all those Bee Gees and Abba LP's?

-MC
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From freeswitch.org at todandlorna.com  Mon Mar  8 15:30:16 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Mon, 08 Mar 2010 16:30:16 -0700
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
Message-ID: <4B958888.7080902@todandlorna.com>

There exists a large number of companies that make large profits largely 
because they sell based on what they think the customer will pay.  NDAs 
keep customers from comparing prices, and competitors from competing on 
price.  This has been a pretty common telecom practice, though not NDA 
specifically.  This is why the "let me see a copy of your phone bill" 
line is so ill-advised to be substantiated when talking to a new 
provider.  If you give them your phone bill, they know exactly where the 
bar must be set to win.

-Tod Hansmann

On 3/8/2010 4:05 PM, Brian West wrote:
> Why on earth do you need an NDA for prices or any info a provider can 
> send you... sounds pretty lame to me...
>
> /b
>
> On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>> And don't forget the propensity of (wholesale) VoIP providers to 
>> require an NDA to be faxed (or, if you're lucky, scanned and 
>> e-mailed) before discussing pricing.
>> Without wishing to digress further, I find this utterly absurd, and 
>> I've yet to find a satisfactory explanation of why it should be.
>> --Dave
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From freeswitch.org at todandlorna.com  Mon Mar  8 15:30:34 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Mon, 08 Mar 2010 16:30:34 -0700
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
Message-ID: <4B95889A.1090907@todandlorna.com>

There are a number of services that do so.  Implementing them wouldn't 
be trivial, but it would be close.

-Tod Hansmann

On 3/8/2010 4:02 PM, Klaus Hochlehnert wrote:
>
> Hi Guys,
>
> just saw that you already have a Facebook entry.
>
> Wouldn't it be possible to connect the news/twitter feeds somehow with 
> Facebook?
>
> I'd really like to see all news on Facebook...
>
> Cheers,
>
> Klaus
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From brian at freeswitch.org  Mon Mar  8 15:34:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 17:34:00 -0600
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <4B95889A.1090907@todandlorna.com>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
Message-ID: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>

Would love to have someone step in and help with these types of things.  As it stands I fight for the time to sleep.  :P

/b

On Mar 8, 2010, at 5:30 PM, Tod Hansmann wrote:

> There are a number of services that do so.  Implementing them wouldn't be trivial, but it would be close.
> 
> -Tod Hansmann




From Mailings at kh-dev.de  Mon Mar  8 15:48:06 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 00:48:06 +0100
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
	<5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4B@srv02.khdev.corp>

I just found a (German) help on how to do this.
Looks like there's a Facebook app that can do this
http://help.twitter.com/entries/86007-erste-schritte-twitter-auf-facebook

If it's as easy as they describe I could help.
I've never done this before, but I'm willing to learn... ;-)


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Tuesday, March 09, 2010 12:34 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Facebook...

Would love to have someone step in and help with these types of things.  As it stands I fight for the time to sleep.  :P

/b

On Mar 8, 2010, at 5:30 PM, Tod Hansmann wrote:

> There are a number of services that do so.  Implementing them wouldn't be trivial, but it would be close.
>
> -Tod Hansmann


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





From anthony.minessale at gmail.com  Mon Mar  8 15:57:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Mar 2010 17:57:25 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <4B958888.7080902@todandlorna.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
Message-ID: <191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>

All these threads seem to go off on on tangents, seems at least a lot of
people are reading it so.....

This thread brought to you by ClueCon MMX!
register today http://www.cluecon.com

Hey, it was worth a shot, maybe we can do a presentation on t38 in FS (if we
ever get the time to finish it).




On Mon, Mar 8, 2010 at 5:30 PM, Tod Hansmann <freeswitch.org at todandlorna.com
> wrote:

>  There exists a large number of companies that make large profits largely
> because they sell based on what they think the customer will pay.  NDAs keep
> customers from comparing prices, and competitors from competing on price.
> This has been a pretty common telecom practice, though not NDA
> specifically.  This is why the "let me see a copy of your phone bill" line
> is so ill-advised to be substantiated when talking to a new provider.  If
> you give them your phone bill, they know exactly where the bar must be set
> to win.
>
> -Tod Hansmann
>
>
> On 3/8/2010 4:05 PM, Brian West wrote:
>
> Why on earth do you need an NDA for prices or any info a provider can send
> you... sounds pretty lame to me...
>
>  /b
>
>  On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>  And don't forget the propensity of (wholesale) VoIP providers to require
> an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before
> discussing pricing.
>
> Without wishing to digress further, I find this utterly absurd, and I've
> yet to find a satisfactory explanation of why it should be.
>
> --Dave
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From steveu at coppice.org  Mon Mar  8 16:34:36 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 09 Mar 2010 08:34:36 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>	<4B951E63.6060200@coppice.org>	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
Message-ID: <4B95979C.7070206@coppice.org>

On 03/09/2010 05:54 AM, Klaus Hochlehnert wrote:
> Hi
>
> you could also use newest trunk of t38modem + trunk of OPAL.
> The newest trunk version(s) use SpanDSP as fax engine and it works more reliable than the previous t38 engine (I think it was integrated in OPAL).
> It also supports fallback to t30 audio fax if you're connected to ISDN network.
>
> Together with HylaFAX+ the success rate is ok (unfortunately not perfect).
>
> Klaus
>    
The success rate with the latest opal + spandsp code should be about as 
about good as your system is capable of giving. That is, most failures 
will either be:

     - that the call is just a bad call - wrong number, messed up FAX 
machine usage, etc.
     - your systems is dropping packets or otherwise screwing up the timing.
     - T.38 negotiation is screwing up, because T.38 compatibility is 
still a mess.

The last one is such a problem for the whole industry, the SIP Forum has 
a working group to address it.

Steve




From steveu at coppice.org  Mon Mar  8 16:38:02 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 09 Mar 2010 08:38:02 +0800
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
Message-ID: <4B95986A.4020708@coppice.org>

On 03/09/2010 07:05 AM, Brian West wrote:
> Why on earth do you need an NDA for prices or any info a provider can 
> send you... sounds pretty lame to me...
>
> /b
Lame? The essence of effective negotiating is to keep all the parties 
involved divided, and each thinking they got a great deal.

"Don't tell the others about this price offer, will you" tends to mean 
"if you talk about these prices, you might find others are paying less".

Steve

>
> On Mar 8, 2010, at 5:01 PM, David Knell wrote:
>
>> And don't forget the propensity of (wholesale) VoIP providers to 
>> require an NDA to be faxed (or, if you're lucky, scanned and 
>> e-mailed) before discussing pricing.
>> Without wishing to digress further, I find this utterly absurd, and 
>> I've yet to find a satisfactory explanation of why it should be.
>> --Dave
>>




From will.traenkle at yahoo.com  Mon Mar  8 16:45:10 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 8 Mar 2010 16:45:10 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
Message-ID: <304682.93065.qm@web57605.mail.re1.yahoo.com>

I did not see this message go through yet so I removed the code I originally had pasted and am resending it.

FS Community,



I am new to freeSWITCH and this mailing list and I appreciate your support in advance.

My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.

The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Ideas? Please advise. 

Thanks.

-Will


      
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From msc at freeswitch.org  Mon Mar  8 16:55:28 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 8 Mar 2010 16:55:28 -0800
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <304682.93065.qm@web57605.mail.re1.yahoo.com>
References: <304682.93065.qm@web57605.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003081655o5e1adf0ap868ec31b0631aab6@mail.gmail.com>

I thought someone responded already and asked that you confirm that you are
ignoring early media in your bridge line...

-MC

On Mon, Mar 8, 2010 at 4:45 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> I did not see this message go through yet so I removed the code I
> originally had pasted and am resending it.
>
> FS Community,
>
> I am new to freeSWITCH and this mailing list and I appreciate your support
> in advance.
>
> My Goal: To create a hunt group x7000 that sequentially dials an internal
> extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while
> playing music the entire time when each number is being tried.
>
> The Issue: This was working great a few months back, but now music plays
> when only when x1000 is tried but NOT when the cell phone is tried.
>
> Ideas? Please advise.
>
> Thanks.
>
> -Will
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Mar  8 17:01:42 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Mar 2010 09:01:42 +0800
Subject: [Freeswitch-users] Which Lua script am I?
In-Reply-To: <5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>
References: <5d2828f1003051104g6011c17w1a669faded390a42@mail.gmail.com>
	<87f2f3b91003051116p524bc237wca60baa697b99789@mail.gmail.com>
	<5d2828f1003051132m7af18875i49567dc0f3a788ce@mail.gmail.com>
	<5d2828f1003051138n5b755291vfcd2c9f5ff34bc08@mail.gmail.com>
	<5084F072-9194-4461-930B-C79AFD978944@freeswitch.org>
	<5d2828f1003051159i48107791m58730e23fe9291e9@mail.gmail.com>
	<5d2828f1003051217g52dd8574t81c9bb349e6a1e63@mail.gmail.com>
	<23f91031003051928j6d0c8f82mbf22483a6c3b06ad@mail.gmail.com>
	<5d2828f1003080900q36f907f5r10044cc295b47608@mail.gmail.com>
Message-ID: <23f91031003081701h4841f7e7x69def77879707117@mail.gmail.com>

The patch was committed on 15433. Though I haven't start to use that.
Which version are you on?
http://jira.freeswitch.org/browse/MODLANG-122

2010/3/9 Mike van Lammeren <mike at van.lammeren.net>:
> Actually, I don't believe that does work. At least, not on my system. Trying
> to access argv[0] just produces a lua error about "concatenating a nil
> value."
>
> On Fri, Mar 5, 2010 at 10:28 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> and also you can get the script name from argv[0] if you name you
>> script in different names
>>
>> 2010/3/6 Mike van Lammeren <mike at van.lammeren.net>:
>> > In case anyone's interested, I just found something cool. You can pass
>> > parameters to the Lua script from the lua.conf.xml file, like so:
>> > ?? ?<param name="startup-script" value="temp.lua script1"/>
>> > ?? ?<param name="startup-script" value="temp.lua script2"/>
>> > ?? ?<param name="startup-script" value="temp.lua script3"/>
>> > I can see that being useful.
>> > Mike van Lammeren
>> >
>> > On Fri, Mar 5, 2010 at 2:59 PM, Mike van Lammeren
>> > <mike at van.lammeren.net>
>> > wrote:
>> >>
>> >> Thanks so much, everyone! Awesome advice, as usual!
>> >> Due to a deadline situation, I'm going with the Lua script idea.
>> >> However,
>> >> if it gives us any trouble after the looming deadline, then I will
>> >> definitely go the C route.
>> >> Thanks again!
>> >> Mike van Lammeren
>> >>
>> >> On Fri, Mar 5, 2010 at 2:42 PM, Brian West <brian at freeswitch.org>
>> >> wrote:
>> >>>
>> >>> Write it in C and use the FreeSWITCH task scheduler API.
>> >>>
>> >>> /b
>> >>>
>> >>> On Mar 5, 2010, at 1:38 PM, Mike van Lammeren wrote:
>> >>>
>> >>> > All the same, I'm still open to suggestions on better ways to
>> >>> > achieve
>> >>> > my ultimate goals!
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From null at invalid.name  Mon Mar  8 17:05:34 2010
From: null at invalid.name (Dan Lane)
Date: Tue, 9 Mar 2010 01:05:34 +0000
Subject: [Freeswitch-users] Facebook...
In-Reply-To: <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
References: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4A@srv02.khdev.corp>
	<4B95889A.1090907@todandlorna.com>
	<5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org>
Message-ID: <b042b23b1003081705r26bf624ep82e732e328cb30f6@mail.gmail.com>

On Mon, Mar 8, 2010 at 11:34 PM, Brian West <brian at freeswitch.org> wrote:
> Would love to have someone step in and help with these types of things. ?As it stands I fight for the time to sleep. ?:P

I could probably spare some time each week looking after social media
/ news stuff for freeswitch if required.

ISTR I've got you on Jabber (null at invalid.name) so grab me off-list to discuss.



From brian at freeswitch.org  Mon Mar  8 17:07:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 19:07:55 -0600
Subject: [Freeswitch-users] SILK Codec
Message-ID: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>

We now have mod_silk in tree it has only been tested on Linux so far.  The silk library itself was released in the IETF draft only an hour and twenty minutes ago.  I wrote a perl script to extract the source from the draft.  Checked in the lib and the codec module which I had written to the binary lib I had a few months ago. 

It has more work to do... but its there if anyone wants to chip in and libtoolize the library, revamp the build system and assist in testing the codec on multiple platforms.

Thanks,
Brian




From dave at 3c.co.uk  Mon Mar  8 17:20:03 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 8 Mar 2010 22:20:03 -0300
Subject: [Freeswitch-users] Attrafax
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com><4B951E63.6060200@coppice.org><4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com><A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp><ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<FDDF0EE896B34D1A80151B6032B1716C@DELL9><2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
Message-ID: <90C16E97D2234FD58E5099C04329ACC3@DELL9>

Hi Tod,

You might be right in certain circumstances, but, in wholesale telecoms, it's pretty easy to get a good idea of where rates for particular destinations are.  Find a collection of ratecards by typing 'wholesale sip termination' into Google and it quickly becomes pretty obvious what's competitive and what isn't.  NDAs are completely superfluous, especially considering that so much buying is done on price.  If I've not got a good reason to want to terminate traffic through you (i.e. you're not AT&T or Verizon or the like), then I'm unlikely to want to go to the trouble of signing an NDA just to see how much your product costs.

--Dave
  ----- Original Message ----- 
  From: Tod Hansmann 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 8:30 PM
  Subject: Re: [Freeswitch-users] Attrafax


  There exists a large number of companies that make large profits largely because they sell based on what they think the customer will pay.  NDAs keep customers from comparing prices, and competitors from competing on price.  This has been a pretty common telecom practice, though not NDA specifically.  This is why the "let me see a copy of your phone bill" line is so ill-advised to be substantiated when talking to a new provider.  If you give them your phone bill, they know exactly where the bar must be set to win.

  -Tod Hansmann

  On 3/8/2010 4:05 PM, Brian West wrote: 
    Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me... 


    /b


    On Mar 8, 2010, at 5:01 PM, David Knell wrote:


      And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

      Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

      --Dave





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From jaybinks at gmail.com  Mon Mar  8 17:27:57 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 11:27:57 +1000
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
Message-ID: <ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>

this is what I love about Freeswitch, and the FS development team.

Well done Brian ( and the rest of the team ).

Jay



On Tue, Mar 9, 2010 at 11:07 AM, Brian West <brian at freeswitch.org> wrote:

> We now have mod_silk in tree it has only been tested on Linux so far.  The
> silk library itself was released in the IETF draft only an hour and twenty
> minutes ago.  I wrote a perl script to extract the source from the draft.
>  Checked in the lib and the codec module which I had written to the binary
> lib I had a few months ago.
>
> It has more work to do... but its there if anyone wants to chip in and
> libtoolize the library, revamp the build system and assist in testing the
> codec on multiple platforms.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From infos at madovsky.org  Mon Mar  8 17:45:54 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 20:45:54 -0500
Subject: [Freeswitch-users] create phone user extension
Message-ID: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>

to create a  phone user extension,
is it need to have his account in conf/directory/default for default.xml ?
if yes so I don't understand why I can register with extension 1003 if there are only 1000-1002 in conf/directory
I use the defaults FS config

Thanks
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From mayamatakeshi at gmail.com  Mon Mar  8 17:59:58 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 9 Mar 2010 10:59:58 +0900
Subject: [Freeswitch-users] create phone user extension
In-Reply-To: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
Message-ID: <15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>

On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:

>  to create a  phone user extension,
> is it need to have his account in conf/directory/default for default.xml ?
>

Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve
directory info.


> if yes so I don't understand why I can register with extension 1003 if
> there are only 1000-1002 in conf/directory
> I use the defaults FS config
>
>
Is the auth-calls parameter set to true in your sofia profile?
<param name="auth-calls" value="true"/>
If not, anyone can register.
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From infos at madovsky.org  Mon Mar  8 18:22:38 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 8 Mar 2010 21:22:38 -0500
Subject: [Freeswitch-users] create phone user extension
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
	<15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>
Message-ID: <338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>

auth-calls are exactly as default install config
<param name="auth-calls" value="$${internal_auth_calls}"/>

for now I have 2 freeeswitch with pgsql DB shared. so does it use mod_xml_odbc implicitly ?
  ----- Original Message ----- 
  From: mayamatakeshi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 08, 2010 8:59 PM
  Subject: Re: [Freeswitch-users] create phone user extension




  On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:

    to create a  phone user extension,
    is it need to have his account in conf/directory/default for default.xml ?

  Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve directory info.
   

    if yes so I don't understand why I can register with extension 1003 if there are only 1000-1002 in conf/directory
    I use the defaults FS config



  Is the auth-calls parameter set to true in your sofia profile?
  <param name="auth-calls" value="true"/>
  If not, anyone can register.



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


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From freeswitch at gilligan.id.au  Mon Mar  8 18:30:34 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 13:30:34 +1100
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com> 
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
Message-ID: <9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>

That decently woks to originate the call but passing that UUID into a new
ManagedSession does not seem to like to the session created using that UUID.
 Once i create a session used the API you suggested how would i then access
the session in the Managed Code?

Chris

On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> You could use the api:
>
>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
> string apiResult = string.Empty;
> string Uuid;
> string NumberToDial = "3475558308";
> string OutgoingCallerID = "2155556666";
>
> Uuid = fsApi.ExecuteString("create_uuid");
>
> apiResult = fsApi.Execute("originate",
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
> Uuid, OutgoingCallerID, NumberToDial));
>
>
>
>
>
>
> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Hi,
>> I am trying to create a mod_managed API application that takes 2 phone
>> numbers as params.  These numbers should then be used to make 2 calls and to
>> bridge them. The issue i am having is working out how to place the first
>> call leg from in mod_managed.  All the methods i have found are for the
>> second leg and require you to pass in the first call leg.
>>
>> I know there are easier way to do this out of the managed code but i want
>> it in the managed code as this is just a proof of concept to prove we can do
>> a callback like system via managed code to be hooked into other external
>> systems.
>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>
>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>> while (s.ready()) {
>> // The call has been answered
>> }
>>
>> This seems to be exactly what i am looking for but seems to be missing
>> from mod_managed. I am hoping someone can tell me how to do something
>> similar in mod_managed since even if it is not part of the managed code
>> mod_managed is meant to have the native access as well so i would assume it
>> would be possible.
>>
>> Thanks in advance
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Mon Mar  8 18:48:08 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 9 Mar 2010 11:48:08 +0900
Subject: [Freeswitch-users] create phone user extension
In-Reply-To: <338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>
References: <FE1FBB6C6319431EB56342FE9F625DD7@MOBILEE1705>
	<15b9404e1003081759u62d43bbbv73bb597479d93d3@mail.gmail.com>
	<338ADF6B976A481D8D191CCD396D98BF@MOBILEE1705>
Message-ID: <15b9404e1003081848q39332c81s5c7a4c1e4f84fa91@mail.gmail.com>

On Tue, Mar 9, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

>  auth-calls are exactly as default install config
> <param name="auth-calls" value="$${internal_auth_calls}"/>
>

Take a look at
/usr/local/freeswitch/log/freeswitch.xml.fsxml
and check what is the value it was resolved at run-time.


> for now I have 2 freeeswitch with pgsql DB shared. so does it use
> mod_xml_odbc implicitly ?
>

No. I just said that directory is not restricted to putting files at
conf/directory
and that you you could use mod_xml_odbc or mod_xml_curl.
But if you are using default FS files, then you are not using them.



>  ----- Original Message -----
> *From:* mayamatakeshi <mayamatakeshi at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 08, 2010 8:59 PM
> *Subject:* Re: [Freeswitch-users] create phone user extension
>
>
> On Tue, Mar 9, 2010 at 10:45 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  to create a  phone user extension,
>> is it need to have his account in conf/directory/default for default.xml ?
>>
>
> Yes, but you can also use mod_xml_curl or mod_xml_odbc etc to serve
> directory info.
>
>
>>  if yes so I don't understand why I can register with extension 1003 if
>> there are only 1000-1002 in conf/directory
>> I use the defaults FS config
>>
>>
> Is the auth-calls parameter set to true in your sofia profile?
> <param name="auth-calls" value="true"/>
> If not, anyone can register.
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nagalenoj at gmail.com  Mon Mar  8 20:33:15 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Mar 2010 10:03:15 +0530
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
In-Reply-To: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
References: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
Message-ID: <fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>

Anything wrong with this question?! I didn't find any replies for this..

On Fri, Mar 5, 2010 at 10:37 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Dear friends,
>      I've faced an issue in event socket. I would want to know why it
> behaves such a way.
>
>      My program is working with the hep of events. So, based on the
> received event, the process will continue it's work. When I need DTMF, I use
> recvEventTimed and in the other cases, I use recvEvent.
>      So, In the mid if caller hangsup, I expect for the SERVER_DISCONNECTED
> event. When the caller hangsup when I'm waiting in recvEvent, I'm getting
> SERVER_DISCONNECTED. But, when I'm waiting in recvEventTimed, I'mnot
> receiving SERVER_DISCONNECTED, instead receiving an undefined value.
>
>     To handle this, I've checked esl connection inside the timeout part and
> put a recvEvent, then I receive SERVER_DISCONNECTED.
>
>     My question is, why am I not receiving SERVER_DISCONNECTED when I recv
> event using recvEventTimed?!
>
> This is a sample program I execute,
>
> require ESL;
> use IO::Socket::INET;
>
> my $ip = "127.0.0.1";
> my $sock = new IO::Socket::INET ( LocalHost => $ip,  LocalPort => '9242',
> Proto => 'tcp',  Listen => 1,  Reuse => 1 );
> die "Could not create socket: $!\n" unless $sock;
>
> for(;;) {
>   my $new_sock = $sock->accept();
>   my $pid = fork();
>   if ($pid) {
>     close($new_sock);
>     next;
>   }
>
>   my $host = $new_sock->sockhost();
>   my $fd = fileno($new_sock);
>
>   my $con = new ESL::ESLconnection($fd);
>   my $info = $con->getInfo();
>   my $uuid = $info->getHeader("unique-id");
>   printf "Connected call %s, from %s\n", $uuid,
> $info->getHeader("caller-caller-id-number");
>
>   my $e = $con->filter("unique-id", $uuid);
>   if ($e) {
>     print $e->serialize();
>   } else {
>     printf("WTF?\n");
>   }
>
>   $con->events("plain", "SERVER_DISCONNECTED DTMF");
>   $con->execute("answer");
>
>   while($con->connected()) {
>     $e = $con->recvEvent();   ############# CASE 1 ##
>     # $e = $con->recvEventTimed(10000);       ############# CASE 2 ##
>
>     unless ($e) {
>         unless ($con->connected()) {
> ###############################
>            $e = $con->recvEvent();                         ##
>            print $e->serialize();                               ##   Added
> code to get SERVER_DISCONNECTED.
>            print "SERVER_DISCONNECTED";     ##
>         }
> ###############################
>         print "DTMF timeout\n";
>     }
>     if ($e) {
>       my $name = $e->getHeader("event-name");
>       print "EVENT [$name]\n";
>       if ($name eq "DTMF") {
>         my $digit = $e->getHeader("dtmf-digit");
>         my $duration = $e->getHeader("dtmf-duration");
>         print "DTMF digit $digit ($duration)\n";
>       }
>     }
>   }
>   print "BYE\n";
>   close($new_sock);
> }
>
> --
> Regards,
> Nagalenoj H.
>



-- 
Regards,
Nagalenoj H.
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From brian at freeswitch.org  Mon Mar  8 20:49:02 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 8 Mar 2010 22:49:02 -0600
Subject: [Freeswitch-users] recvEventTimed - SERVER_DISCONNECTED
In-Reply-To: <fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>
References: <fe67a3341003042107u35192e8eoadf5833b0938bcb4@mail.gmail.com>
	<fe67a3341003082033l634b6e9t3bbba59054e3370b@mail.gmail.com>
Message-ID: <D19C6489-26D8-4AD3-A5EF-E2D6436A13F6@freeswitch.org>

This is because you're timing out and ending your loop and loosing the event.   while($con->connected()) { becomes false and you can't come back around to pickup the event because you would have been disconnected before you had a chance to get the event.   recvEventTimed allows you to do other things if you don't receive events in a set amount of time.

One example of proper usage on inbound event socket:

http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/ESL/Dispatch.pm
http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/dispatch.pl


/b
On Mar 8, 2010, at 10:33 PM, Nagalenoj H. wrote:

> Anything wrong with this question?! I didn't find any replies for this..




From infos at madovsky.org  Mon Mar  8 21:58:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 00:58:10 -0500
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
Message-ID: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>

I noticed that if a registration doesn't close properly and
try to reconnect I can see in the table sip_registrations the old reg and new one.
is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?

Thanks

Franck
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From brian at freeswitch.org  Mon Mar  8 22:13:24 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 00:13:24 -0600
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
In-Reply-To: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
References: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
Message-ID: <76633F3B-93B1-47CF-A3E8-FDD07FD817D3@freeswitch.org>

old reg? what do you mean and how do you mean ?  and the hostname is how it tells.
On Mar 8, 2010, at 11:58 PM, Madovsky wrote:

> I noticed that if a registration doesn't close properly and
> try to reconnect I can see in the table sip_registrations the old reg and new one.
> is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?
>  
> Thanks
>  
> Franck
> _______

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From jbrucehopkins at gmail.com  Mon Mar  8 22:22:52 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 9 Mar 2010 06:22:52 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
Message-ID: <63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>

Ah I see.  I will try again using the internal profile and forwarding  
port 5060.  Presumably still creating a directory entry to enable the  
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two  
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing  
>> something wrong here as I am still having difficulty, and still  
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding  
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to  
>> retain an entry in the directory /usr/local/freeeswitch/conf/ 
>> directory/ext_dns.xml in order to give freeswitch the dns name of  
>> the server as a domain for the remote softphone to register on.  I  
>> left the group name in this entry the same as inthe default entry,  
>> so that the remote phone could register on the same extension  
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN  
>> as freeswitch) phone to the remote phone, I get the message on the  
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of  
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment  
>> <param name="aggressive-nat-detection" value="true"/> in  
>> external.xml, otherwise in the case of a call initiated by the  
>> remote phone being hung up by the local phone, freeswitch sent the  
>> BYE to the private IP of the remote phone, rather than its public  
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From infos at madovsky.org  Mon Mar  8 22:29:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 01:29:18 -0500
Subject: [Freeswitch-users] sip registrations and postgresql odbc-dsn
References: <FA20DE66D6C84CB083B3B23C31555131@MOBILEE1705>
	<76633F3B-93B1-47CF-A3E8-FDD07FD817D3@freeswitch.org>
Message-ID: <05F13F0A7E3B40C78F8BB4BC1061032A@MOBILEE1705>

I mean if legA register on host1, suddenly FS or the client shutdown and register again on host1 or host2 so
in pgsql the old call_id row exsits yet, and a new row with the ne call_id  is created...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:13 AM
  Subject: Re: [Freeswitch-users] sip registrations and postgresql odbc-dsn


  old reg? what do you mean and how do you mean ?  and the hostname is how it tells.

  On Mar 8, 2010, at 11:58 PM, Madovsky wrote:


    I noticed that if a registration doesn't close properly and
    try to reconnect I can see in the table sip_registrations the old reg and new one.
    is it normal ? if yes, How FS can guess the right registration in case of multiple FS with shared DB ?

    Thanks

    Franck
    _______




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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar  8 23:59:40 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 02:59:40 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
Message-ID: <663CE110CAE64F77BE2FEF42B1905A2D@MOBILEE1705>

in sip_registration table there are
call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname 
fields, so how can I get these vars from the callee before the bridge ?
I need to replace the host of the callee in case of if he's registered on another FS node.

thanks

Franck
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From infos at madovsky.org  Tue Mar  9 00:04:22 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 03:04:22 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
Message-ID: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>

forgot to say that I need it (since I didn't find any good examples)
for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:59 AM
  Subject: vars from sip_registrations table


  in sip_registration table there are
  call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname 
  fields, so how can I get these vars from the callee before the bridge ?
  I need to replace the host of the callee in case of if he's registered on another FS node.

  thanks

  Franck
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From will.traenkle at yahoo.com  Tue Mar  9 00:47:50 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 00:47:50 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <20100308184025.GL18427@base.carmickle.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
	<20100308184025.GL18427@base.carmickle.com>
Message-ID: <358354.17194.qm@web57606.mail.re1.yahoo.com>

Thanks Frank.  I will try that out.

Cheers,




________________________________
From: Frank Carmickle <frank at carmickle.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, March 8, 2010 10:40:25 AM
Subject: Re: [Freeswitch-users] hunt group - play music when trying external numbers

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC

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



      
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From siniypin at gmail.com  Tue Mar  9 01:17:23 2010
From: siniypin at gmail.com (RobertT)
Date: Tue, 9 Mar 2010 12:17:23 +0300
Subject: [Freeswitch-users] ICE related sdp headers forwarding
Message-ID: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>

Hi guys!

In my installation I have a public FS being connected by lots of clients and
being used to connect them with each other. Some of them might be backed by
various NATs. And though they all use STUN to discover their public IP
address, sometimes it's just not enough to provide two-way sound. These
clients all have ICE and TURN support, but it is useless, since FS doesn't
support ICE (at least that's what I know), and there are no ICE-related
headers in FS's originating SDP for b-leg. Currently, all calls are being
connected by means of bridge app with proxy_media mode enabled.

For now, I can only think of couple of ways to enable ICE negotiation
between a and b:
1. Redirect a directly to b. But to accomplish this I've to know b's IP,
that it was registered with, which is being stored in sip_registrations
db. Can I obtain it with db app in dialplan and substitute as a parameter to
deflect app?
2. Force FS to copy remote SDP headers, or at least a part of them, to it's
local SDP.
3. I feel I might be missing something important...

Please help me to find a best solution to enable ICE negotiation.

Best regards, RobertT.
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From yehavi.bourvine at gmail.com  Tue Mar  9 01:45:24 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 11:45:24 +0200
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
Message-ID: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>

Is there a simple way/command inside FreeSwitch to send a NOTIFY message to
a Polycom phone which tells it to reboot? I know it is possible but I do not
find any documentation about it...

                     Thanks, __Yehavi:
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From jaybinks at gmail.com  Tue Mar  9 02:08:13 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 20:08:13 +1000
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
In-Reply-To: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
References: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
Message-ID: <ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>

this should help you on your way.

http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints

<http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints>
Jay

On Tue, Mar 9, 2010 at 7:45 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Is there a simple way/command inside FreeSwitch to send a NOTIFY message to
> a Polycom phone which tells it to reboot? I know it is possible but I do not
> find any documentation about it...
>
>                      Thanks, __Yehavi:
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From freeswitch at gilligan.id.au  Tue Mar  9 02:27:31 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Tue, 9 Mar 2010 21:27:31 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
Message-ID: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>

What i need to do is listen on a session before, during and after the
session is bridged with another call. I know built into FS is the option to
listen for *x where x is 0-9 but i really need more than one digit.

I traced the method that listens for the single digit to the
command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
switch_ivr_async.c

mod_managed also has the following in the native class freeswitch.
public static switch_status_t
switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
recv_dtmf)

The problem is i can't work out how to use it.  I am fine the the first
param for the session but have no idea what i need for
the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
recv_dtmf param.

if anyone can point me in the right direction or knows a better way i would
be grateful.

Chris
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From yehavi.bourvine at gmail.com  Tue Mar  9 03:13:16 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 13:13:16 +0200
Subject: [Freeswitch-users] Sending SIP notify to polycom in order to
	resync/reboot it
In-Reply-To: <ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>
References: <d361ca7b1003090145o1b1e1bf2l803a168fc983c95c@mail.gmail.com>
	<ec2ce17e1003090208x7b640f1eo7b17d83c81d67de8@mail.gmail.com>
Message-ID: <d361ca7b1003090313x217e7ac8n686a687285a127d6@mail.gmail.com>

Thanks!
                 __Yehavi:

 2010/3/9 jay binks <jaybinks at gmail.com>

> this should help you on your way.
>
>
> http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints
>
> <http://wiki.freeswitch.org/wiki/Mod_sofia#Flushing.2Frebooting_registered_endpoints>
> Jay
>
>  On Tue, Mar 9, 2010 at 7:45 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>>  Is there a simple way/command inside FreeSwitch to send a NOTIFY message
>> to a Polycom phone which tells it to reboot? I know it is possible but I do
>> not find any documentation about it...
>>
>>                      Thanks, __Yehavi:
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Stephen.Kingham at kingtech.com.au  Tue Mar  9 00:18:39 2010
From: Stephen.Kingham at kingtech.com.au (Stephen Kingham)
Date: Tue, 09 Mar 2010 19:18:39 +1100
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <4B9532AB.5030407@telemaque.fr>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com>
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com>
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr>
Message-ID: <1268122719.27572.2.camel@zen>

Hi

I can help out with the Cisco config, but I have not worked out what I
need to configure FreeSWITCH to route a SIP call to the ip address of
the Cisco GW.  If I could get a bit of help there I will write, or help
write howto.

Regards

Stevek


On Mon, 2010-03-08 at 18:23 +0100, Tristan Mah? wrote:
> +1 with Brian,
> 
> Just as I have 5 spare minutes and to be kind:
> 
> 
> * E1 controller config:
> 
> controller E1 0/0/0
>  framing NO-CRC4
>  pri-group timeslots 1-31
>  description E1_1
> !
> interface Serial0/0/0:15
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-net5
>  isdn not-end-to-end 64
>  isdn incoming-voice voice
>  isdn bchan-number-order descending round-robin
>  no cdp enable
> !
> voice-port 0/0/0:15
>  no comfort-noise
>  cptone FR
>  timeouts call-disconnect 1
>  timeouts wait-release 1
> !
> 
> * Dial-peers ( SIP to E1 and E1 to SIP):
> 
> dial-peer voice 100 pots
>  description SIP_TO_E1
>  destination-pattern 0........
>  progress_ind setup enable 3
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  progress_ind disconnect enable 8
>  no digit-strip
>  port 0/0/0:15
> !
> dial-peer voice 500 voip
>  description E1_TO_SIP
>  destination-pattern ....
>  progress_ind setup enable 3
>  session protocol sipv2
>  session target ipv4:FREESWITCH_IP
>  session transport udp
>  dtmf-relay rtp-nte
>  codec g711alaw
>  no vad
> !
> 
> 
> Remember to adapt this config to your needs ( there are mandatory
> missing parts, but at least you have some config options to look for).
> 
> Regards,
> 
> Gled.
> 
> Brian West a ?crit :
> > You seem rather needy.  Have you even tried google yet?  When asking for help you could be more descriptive if you want help.  These "I need help" without any directed questioned signals to me you have a job to do and you don't know how to do it and you're now asking US to do your job for you.   Be descriptive and do some research yourself.  Again we aren't cisco and you didn't pay us for that hardware so I would start by venturing over to www.cisco.com and searching their docs for that hardware on how to configure voice peers.  Then ask questions if anything is left to question.
> > 
> > /b
> > 
> > On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> > 
> >> HI,
> >>  
> >> how do i configure freeswitch gateway to communicate to cisco router, i am using cisco2921.
> >>  
> >> Thanks
> >> Srinivas
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jaybinks at gmail.com  Tue Mar  9 04:47:05 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 9 Mar 2010 22:47:05 +1000
Subject: [Freeswitch-users] Freeswitch to cisco router(gateway)
In-Reply-To: <1268122719.27572.2.camel@zen>
References: <f8af5741003080708v17b57bb5u8b71fb719009fa9@mail.gmail.com> 
	<f8af5741003080723s3a693d91vbb83efcfd28204f4@mail.gmail.com> 
	<9BEDBC36-4830-4682-94C4-15085B150005@freeswitch.org>
	<4B9532AB.5030407@telemaque.fr> <1268122719.27572.2.camel@zen>
Message-ID: <ec2ce17e1003090447l78eef33x643e84289d3fb2d8@mail.gmail.com>

all you need is..

<action application="bridge"
data="{sip_cid_type=rpid}sofia/**YOURPROFILE**/$1@**CISCOIP**" />

not all IOS's are going to need the caller ID to be forced to rpid,
but in my experience I needed it.  ( it should not hurt anyways )

hope that helps

Jay

On Tue, Mar 9, 2010 at 6:18 PM, Stephen Kingham <
Stephen.Kingham at kingtech.com.au> wrote:

> Hi
>
> I can help out with the Cisco config, but I have not worked out what I
> need to configure FreeSWITCH to route a SIP call to the ip address of
> the Cisco GW.  If I could get a bit of help there I will write, or help
> write howto.
>
> Regards
>
> Stevek
>
>
> On Mon, 2010-03-08 at 18:23 +0100, Tristan Mah? wrote:
> > +1 with Brian,
> >
> > Just as I have 5 spare minutes and to be kind:
> >
> >
> > * E1 controller config:
> >
> > controller E1 0/0/0
> >  framing NO-CRC4
> >  pri-group timeslots 1-31
> >  description E1_1
> > !
> > interface Serial0/0/0:15
> >  no ip address
> >  encapsulation hdlc
> >  isdn switch-type primary-net5
> >  isdn not-end-to-end 64
> >  isdn incoming-voice voice
> >  isdn bchan-number-order descending round-robin
> >  no cdp enable
> > !
> > voice-port 0/0/0:15
> >  no comfort-noise
> >  cptone FR
> >  timeouts call-disconnect 1
> >  timeouts wait-release 1
> > !
> >
> > * Dial-peers ( SIP to E1 and E1 to SIP):
> >
> > dial-peer voice 100 pots
> >  description SIP_TO_E1
> >  destination-pattern 0........
> >  progress_ind setup enable 3
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  progress_ind disconnect enable 8
> >  no digit-strip
> >  port 0/0/0:15
> > !
> > dial-peer voice 500 voip
> >  description E1_TO_SIP
> >  destination-pattern ....
> >  progress_ind setup enable 3
> >  session protocol sipv2
> >  session target ipv4:FREESWITCH_IP
> >  session transport udp
> >  dtmf-relay rtp-nte
> >  codec g711alaw
> >  no vad
> > !
> >
> >
> > Remember to adapt this config to your needs ( there are mandatory
> > missing parts, but at least you have some config options to look for).
> >
> > Regards,
> >
> > Gled.
> >
> > Brian West a ?crit :
> > > You seem rather needy.  Have you even tried google yet?  When asking
> for help you could be more descriptive if you want help.  These "I need
> help" without any directed questioned signals to me you have a job to do and
> you don't know how to do it and you're now asking US to do your job for you.
>   Be descriptive and do some research yourself.  Again we aren't cisco and
> you didn't pay us for that hardware so I would start by venturing over to
> www.cisco.com and searching their docs for that hardware on how to
> configure voice peers.  Then ask questions if anything is left to question.
> > >
> > > /b
> > >
> > > On Mar 8, 2010, at 9:23 AM, srinivasula reddy wrote:
> > >
> > >> HI,
> > >>
> > >> how do i configure freeswitch gateway to communicate to cisco router,
> i am using cisco2921.
> > >>
> > >> Thanks
> > >> Srinivas
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely

Jay
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From srinivas.ksvreddy at gmail.com  Tue Mar  9 05:47:23 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 9 Mar 2010 19:17:23 +0530
Subject: [Freeswitch-users] freeswitch to cisco router
Message-ID: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>

Hi,

i have tried to configure gateway from my freeswitch, with the following
credentials proxy(ip of cisco) and username,password.
and when i start my freeswitch registration reqest is going to cisco router
and again it requesting to same registration reqeust to my sipserver, the
register packet from cisco router to my freeswitch is comming like this

To: sip:.*@sipserverip:5060

so there there is no user present with the name of .* it is rejecting the
registration request, any idea

Thanks
Swapna
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From yehavi.bourvine at gmail.com  Tue Mar  9 06:24:39 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 9 Mar 2010 16:24:39 +0200
Subject: [Freeswitch-users] freeswitch to cisco router
In-Reply-To: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
References: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
Message-ID: <d361ca7b1003090624p202563e2k434d9aabe7a26559@mail.gmail.com>

Cisco gateways usually do not register with the server (thus you have to
allow them via ACL). If you post your config we might help.

                 __Yehavi:

 2010/3/9 srinivasula reddy <srinivas.ksvreddy at gmail.com>

>  Hi,
>
> i have tried to configure gateway from my freeswitch, with the following
> credentials proxy(ip of cisco) and username,password.
> and when i start my freeswitch registration reqest is going to cisco router
> and again it requesting to same registration reqeust to my sipserver, the
> register packet from cisco router to my freeswitch is comming like this
>
> To: sip:.*@sipserverip:5060
>
> so there there is no user present with the name of .* it is rejecting the
> registration request, any idea
>
> Thanks
> Swapna
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Tue Mar  9 07:43:10 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 09:43:10 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B955401.1030004@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
	<4B955401.1030004@marketelsystems.com>
Message-ID: <4B966C8E.9040805@marketelsystems.com>

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From anthony.minessale at gmail.com  Tue Mar  9 08:39:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 10:39:17 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
Message-ID: <191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>

You would need to make a patch for it or pay someone to add it.


On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

>  forgot to say that I need it (since I didn't find any good examples)
> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
> not compatible)
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 2:59 AM
> *Subject:* vars from sip_registrations table
>
> in sip_registration table there are
> call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>
> fields, so how can I get these vars from the callee before the bridge ?
> I need to replace the host of the callee in case of if he's registered on
> another FS node.
>
> thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  9 08:47:35 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 11:47:35 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <F996975CC7C34E28A54D55CEBC5E4CB0@MOBILEE1705>

Ok it's what I guessed.

Thanks
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Tue Mar  9 08:51:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 10:51:30 -0600
Subject: [Freeswitch-users] Eavesdrop
In-Reply-To: <4B966C8E.9040805@marketelsystems.com>
References: <4B95244A.8030808@marketelsystems.com>
	<b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com>
	<4B955401.1030004@marketelsystems.com>
	<4B966C8E.9040805@marketelsystems.com>
Message-ID: <191c3a031003090851m66f89d6ey9e20cb8d8f064852@mail.gmail.com>

you are missing:

call-command: execute

and optionally:

event-lock: true // if you want this event to be exclusive until its done
and
async: true // if you are in sync mode and you want to do the command async


you might want to look at the ESL lib and use that instead of rolling your
own event socket code which
usually generates many question and answer sessions like this one.




-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  9 08:55:29 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 11:55:29 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<F996975CC7C34E28A54D55CEBC5E4CB0@MOBILEE1705>
Message-ID: <EE8802DA199140B581FACD5359022E9A@MOBILEE1705>

I Have no money, but a nice Gibson Les Paul standard from 1990 to offer if
my experimental project will end soon...
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:47 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Ok it's what I guessed.

  Thanks
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 09, 2010 11:39 AM
    Subject: Re: [Freeswitch-users] vars from sip_registrations table


    You would need to make a patch for it or pay someone to add it.



    On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

      forgot to say that I need it (since I didn't find any good examples)
      for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

      Thanks

      F
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 2:59 AM
        Subject: vars from sip_registrations table


        in sip_registration table there are
              call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

        fields, so how can I get these vars from the callee before the bridge ?
        I need to replace the host of the callee in case of if he's registered on another FS node.

        thanks

        Franck

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





    -- 
    Anthony Minessale II

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

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

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



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From dave.redmore at spigotsystems.com  Tue Mar  9 08:44:03 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Tue, 9 Mar 2010 10:44:03 -0600 (CST)
Subject: [Freeswitch-users] Freeswitch a good fit?
In-Reply-To: <22405606.601268153039310.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>

Hi All, 

New to the list and just starting to look at Freeswitch. 

I have an application that I'm looking at and wondering if Freeswitch is the right fit. 

I need a (fairly) simple SIP server and a T.38 media proxy. The setup would be SIP FXS units at the customers location doing outbound faxing. via T.38. The FXS needs to register with my server, which will authenticate the user and connect the call to my T.38 provider's gateway. 

I've seen some reference to using a "proxy-media" setting in the WIKI, but refers to it being a "hack" - so not a big confidence builder. 

I've also looked at using OpenSIPS and MediaProxy, which looks to do what I'm trying to accomplish as well. 

Does it make sense to deploy FS in this scenario? 

Thanks, 

Dave Redmore 
Spigot Networks, Inc. 
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From infos at madovsky.org  Tue Mar  9 09:37:03 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:37:03 -0500
Subject: [Freeswitch-users] Eavesdrop
References: <4B95244A.8030808@marketelsystems.com><b39273531003080855t4b32095cwaa12fee2dc66f3b7@mail.gmail.com><4B955401.1030004@marketelsystems.com><4B966C8E.9040805@marketelsystems.com>
	<191c3a031003090851m66f89d6ey9e20cb8d8f064852@mail.gmail.com>
Message-ID: <BD0D7EF57A5142B7B28B7D430F5A1DEA@MOBILEE1705>

also I would love to make a patch, but i"m really not C or C++ guru...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:51 AM
  Subject: Re: [Freeswitch-users] Eavesdrop


  you are missing:

  call-command: execute

  and optionally:

  event-lock: true // if you want this event to be exclusive until its done
  and
  async: true // if you are in sync mode and you want to do the command async


  you might want to look at the ESL lib and use that instead of rolling your own event socket code which
  usually generates many question and answer sessions like this one.




  -- 
  Anthony Minessale II

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

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

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



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From Mailings at kh-dev.de  Tue Mar  9 09:36:54 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 9 Mar 2010 18:36:54 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<FDDF0EE896B34D1A80151B6032B1716C@DELL9>
	<2650A49F-DEC5-4A83-8FA5-7A52B9156091@freeswitch.org>
	<4B958888.7080902@todandlorna.com>
	<191c3a031003081557j17de448dmbc8253f1dfd45f80@mail.gmail.com>
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA4D@srv02.khdev.corp>

Good idea...

But why don't you send a fax to everyone...   ;-)


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Tuesday, March 09, 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Attrafax

All these threads seem to go off on on tangents, seems at least a lot of people are reading it so.....

This thread brought to you by ClueCon MMX!
register today http://www.cluecon.com

Hey, it was worth a shot, maybe we can do a presentation on t38 in FS (if we ever get the time to finish it).



On Mon, Mar 8, 2010 at 5:30 PM, Tod Hansmann <freeswitch.org<http://freeswitch.org>@todandlorna.com<http://todandlorna.com>> wrote:
There exists a large number of companies that make large profits largely because they sell based on what they think the customer will pay.  NDAs keep customers from comparing prices, and competitors from competing on price.  This has been a pretty common telecom practice, though not NDA specifically.  This is why the "let me see a copy of your phone bill" line is so ill-advised to be substantiated when talking to a new provider.  If you give them your phone bill, they know exactly where the bar must be set to win.

-Tod Hansmann


On 3/8/2010 4:05 PM, Brian West wrote:
Why on earth do you need an NDA for prices or any info a provider can send you... sounds pretty lame to me...

/b

On Mar 8, 2010, at 5:01 PM, David Knell wrote:


And don't forget the propensity of (wholesale) VoIP providers to require an NDA to be faxed (or, if you're lucky, scanned and e-mailed) before discussing pricing.

Without wishing to digress further, I find this utterly absurd, and I've yet to find a satisfactory explanation of why it should be.

--Dave





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

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  9 09:37:49 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:37:49 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <DABE813E4C5D4AA58E163ECB00355A24@MOBILEE1705>

soooorry I answered to the wrong thread in my last email
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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  -- 
  Anthony Minessale II

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

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

  FreeSWITCH Developer Conference
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  iax:guest at conference.freeswitch.org/888
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  pstn:+19193869900



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From infos at madovsky.org  Tue Mar  9 09:39:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 12:39:18 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
Message-ID: <509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>

OK How much ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 11:39 AM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  You would need to make a patch for it or pay someone to add it.



  On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say that I need it (since I didn't find any good examples)
    for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 2:59 AM
      Subject: vars from sip_registrations table


      in sip_registration table there are
            call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

      fields, so how can I get these vars from the callee before the bridge ?
      I need to replace the host of the callee in case of if he's registered on another FS node.

      thanks

      Franck

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





  -- 
  Anthony Minessale II

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

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

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



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


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From anthony.minessale at gmail.com  Tue Mar  9 09:47:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 11:47:24 -0600
Subject: [Freeswitch-users] Freeswitch a good fit?
In-Reply-To: <6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>
References: <22405606.601268153039310.JavaMail.root@zimbra1.spigotsystems.com>
	<6972445.621268153043224.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <191c3a031003090947r384d5b05ve9e534df1db8ea2@mail.gmail.com>

Its not a hack in the sense that it's not likely to work.  It's a hack in
the sense that FS acts mostly as a b2bua role and proxy is a feature mostly
left to proxy software.

I would setup both and compare, using FS might be overkill if you are not
planning to ever do anything else besides proxy t38.



On Tue, Mar 9, 2010 at 10:44 AM, Dave Redmore <
dave.redmore at spigotsystems.com> wrote:

> Hi All,
>
> New to the list and just starting to look at Freeswitch.
>
> I have an application that I'm looking at and wondering if Freeswitch is
> the right fit.
>
> I need a (fairly) simple SIP server and a T.38 media proxy.  The setup
> would be SIP FXS units at the customers location doing outbound faxing. via
> T.38.  The FXS needs to register with my server, which will authenticate the
> user and connect the call to my T.38 provider's gateway.
>
> I've seen some reference to using a "proxy-media" setting in the WIKI, but
> refers to it being a "hack" - so not a big confidence builder.
>
> I've also looked at using OpenSIPS and MediaProxy, which looks to do what
> I'm trying to accomplish as well.
>
> Does it make sense to deploy FS in this scenario?
>
> Thanks,
>
> Dave Redmore
> Spigot Networks, Inc.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From sekharreddypk at gmail.com  Tue Mar  9 09:40:37 2010
From: sekharreddypk at gmail.com (chandrasekhar reddy)
Date: Tue, 9 Mar 2010 23:10:37 +0530
Subject: [Freeswitch-users] call forward
Message-ID: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>

Hi,

i am trying to forward call from freeswitch to cisco router, call is
estabilished but after 30sec call got ended, any idea

sekhar
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From sean at obscuradigital.com  Tue Mar  9 09:59:18 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 09 Mar 2010 09:59:18 -0800
Subject: [Freeswitch-users] Outbound calls mobile
Message-ID: <C7BBCC76.5FEF%sean@obscuradigital.com>

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean
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From andrew at hijacked.us  Tue Mar  9 10:06:00 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 9 Mar 2010 13:06:00 -0500
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBCC76.5FEF%sean@obscuradigital.com>
References: <C7BBCC76.5FEF%sean@obscuradigital.com>
Message-ID: <20100309180600.GG18802@hijacked.us>

On Tue, Mar 09, 2010 at 09:59:18AM -0800, Sean Holt wrote:
> Hello, having weird issue with outbound calls to any mobile phone, but not
> to LAN/analog lines.  Curious if anyone else has had this experienced with
> their freeswitch setup.
> 
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
>
Make sure the CallerID you're sending isn't making the provider reject
it. I've seen this happen several times.

Andrew



From peder at networkoblivion.com  Tue Mar  9 10:08:35 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Mar 2010 12:08:35 -0600
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBCC76.5FEF%sean@obscuradigital.com>
References: <C7BBCC76.5FEF%sean@obscuradigital.com>
Message-ID: <107001cabfb3$891cd5e0$9b5681a0$@com>

What are the problems?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 11:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Outbound calls mobile

 

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean 

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From anthony.minessale at gmail.com  Tue Mar  9 10:10:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 12:10:42 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
Message-ID: <191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>

Aww, The les paul sounded nice I have one on my wishlist =D
http://bit.ly/bRHKlk

*shrug* $250.00 maybe?


On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

>  OK How much ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 11:39 AM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> You would need to make a patch for it or pay someone to add it.
>
>
> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  forgot to say that I need it (since I didn't find any good examples)
>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>> not compatible)
>>
>> Thanks
>>
>> F
>>
>>  ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>> *Subject:* vars from sip_registrations table
>>
>> in sip_registration table there are
>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>
>> fields, so how can I get these vars from the callee before the bridge ?
>> I need to replace the host of the callee in case of if he's registered on
>> another FS node.
>>
>> thanks
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar  9 10:25:40 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 13:25:40 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
Message-ID: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>

I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
I understand perfectly your mind of how to manage an open source project, but don't forget
that there are guys like you on this emailist who don't even make any decent salary, that is why
I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
but not                                         share -> ask money -> make people weird -> money won't come
it's my point of view.

Regards
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:10 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Aww, The les paul sounded nice I have one on my wishlist =D 
  http://bit.ly/bRHKlk

  *shrug* $250.00 maybe?



  On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

    OK How much ?
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 11:39 AM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      You would need to make a patch for it or pay someone to add it.



      On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

        forgot to say that I need it (since I didn't find any good examples)
        for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

        Thanks

        F
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 2:59 AM
          Subject: vars from sip_registrations table


          in sip_registration table there are
                call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

          fields, so how can I get these vars from the callee before the bridge ?
          I need to replace the host of the callee in case of if he's registered on another FS node.

          thanks

          Franck

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





      -- 
      Anthony Minessale II

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

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

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



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


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


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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sean at obscuradigital.com  Tue Mar  9 10:30:26 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 09 Mar 2010 10:30:26 -0800
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <107001cabfb3$891cd5e0$9b5681a0$@com>
Message-ID: <C7BBD3C2.5FF6%sean@obscuradigital.com>


Basically I have sip trace turned on and I?m watching the call.  It gets to
the point where it is ?Trying? to esbstalish the call then for no reason it
times out.  No further details from the sip trace other then call has timed
out.

The strange part is this doesn?t happen when calling any analog line.

If you want I can provide the sip trace.

Sean


On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:

> What are the problems?
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Holt
> Sent: Tuesday, March 09, 2010 11:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Outbound calls mobile
>  
> Hello, having weird issue with outbound calls to any mobile phone, but not to
> LAN/analog lines.  Curious if anyone else has had this experienced with their
> freeswitch setup.
> 
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
> 
> Thanks
> Sean 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Mar  9 10:45:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 12:45:02 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
Message-ID: <191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>

ORLY,

So maybe if you *think* you know what you are talking about, you should go
do some research on the average amount of code I write on a daily basis
nonstop for the last several years without "ask money".  This includes most
of the core, and 90% of the modules you are playing with right now including
the 24 thousand lines of code in mod_sofia alone.
So frankly, since I don't see an urgent need for your request, and I am too
busy to do it, this particular feature would require a bounty to complete.

By the way, you have sent like 2 dozen emails in the same week asking for
all kinds of help, that's also being shared to you for free so I think you
are getting the better side of the bargain here.  Asking for a bounty on a
feature that only helps one individual is very common.............




On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

>  I said I OFFER it in case of I finish my (no money no profit but big
> potential) experimental project.
> I understand perfectly your mind of how to manage an open source project,
> but don't forget
> that there are guys like you on this emailist who don't even make any
> decent salary, that is why
> I use and develop open source : share -> evovle your minds -> make all
> people happy -> money comes
> but not                                         share -> ask money -> make
> people weird -> money won't come
> it's my point of view.
>
> Regards
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 1:10 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Aww, The les paul sounded nice I have one on my wishlist =D
> http://bit.ly/bRHKlk
>
> *shrug* $250.00 maybe?
>
>
> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  OK How much ?
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>  *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> You would need to make a patch for it or pay someone to add it.
>>
>>
>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  forgot to say that I need it (since I didn't find any good examples)
>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>>> not compatible)
>>>
>>> Thanks
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>> *Subject:* vars from sip_registrations table
>>>
>>> in sip_registration table there are
>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>
>>> fields, so how can I get these vars from the callee before the bridge ?
>>> I need to replace the host of the callee in case of if he's registered on
>>> another FS node.
>>>
>>> thanks
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From peder at networkoblivion.com  Tue Mar  9 10:47:32 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 9 Mar 2010 12:47:32 -0600
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <C7BBD3C2.5FF6%sean@obscuradigital.com>
References: <107001cabfb3$891cd5e0$9b5681a0$@com>
	<C7BBD3C2.5FF6%sean@obscuradigital.com>
Message-ID: <10e701cabfb8$fa211440$ee633cc0$@com>

Is this out a sip trunk, or a PRI, or some other gateway?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 12:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound calls mobile

 


Basically I have sip trace turned on and I'm watching the call.  It gets to
the point where it is "Trying" to esbstalish the call then for no reason it
times out.  No further details from the sip trace other then call has timed
out.

The strange part is this doesn't happen when calling any analog line.

If you want I can provide the sip trace.

Sean


On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:

What are the problems?
 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, March 09, 2010 11:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Outbound calls mobile

Hello, having weird issue with outbound calls to any mobile phone, but not
to LAN/analog lines.  Curious if anyone else has had this experienced with
their freeswitch setup.

My setup:
Centos 5.4 32bit
Freeswitch 1.0.5 most recent svn trunk

Thanks
Sean 

  _____  

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

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From msc at freeswitch.org  Tue Mar  9 11:01:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 11:01:31 -0800
Subject: [Freeswitch-users] auto attendant -> hunt group -> external
	number does not ring
In-Reply-To: <302537.41702.qm@web57609.mail.re1.yahoo.com>
References: <302537.41702.qm@web57609.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>

On Mon, Mar 8, 2010 at 2:23 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> FS Community,
>
>
> I created an extension 704
>
>
>
> I created a hunt group 7004 that first rings 704 then rings my cell phone
> 16194548924.
>
>
>
> I created an auto attendant 5002 with an option of 704 that goes to hunt
> group 7004.
>
>
>
> 1)     What does work: When I call the hunt group directly by dialing 7004
> internally, the hunt group first rings 704 then rings my cell phone
> successfully.  This is perfect.
>
>
>
> 2)     What does not work: When I call the main auto attendant from either
> internally by dialing 5002 or externally by dialing my main number, and dial
> 704, my extension rings and after not answering my cell phone does not ring.
>
>
>
> What I am expecting in #2 is for my extension 704 to ring and if I do not
> pick up then my cell phone will ring just like it does in #1.
>
>
>
> My knowledge is limited in this area and if you could point me in the right
> direction, that would be great.  I a have tried everything but with no
> luck.
>
>
> Pastebin the necessary debug info. Check
http://wiki.freeswitch.org/wiki/Reporting_Bugs for specifics on how to do
collect and report all the data to pastebin. It would be good to see the
dialplan and the debug log for each call.

When you've collected all the data please reply here with the pastebin URL.
-MC
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From anthony.minessale at gmail.com  Tue Mar  9 11:06:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 13:06:07 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>

Also, when you are done insulting me, maybe try sofia_contact api command
that will resolve your destination to the exact contact string that the
phone registered with which is really the only bit of info
that you would possibly need rather than asking for another expensive
database lookup per call.


On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ORLY,
>
> So maybe if you *think* you know what you are talking about, you should go
> do some research on the average amount of code I write on a daily basis
> nonstop for the last several years without "ask money".  This includes most
> of the core, and 90% of the modules you are playing with right now including
> the 24 thousand lines of code in mod_sofia alone.
> So frankly, since I don't see an urgent need for your request, and I am too
> busy to do it, this particular feature would require a bounty to complete.
>
> By the way, you have sent like 2 dozen emails in the same week asking for
> all kinds of help, that's also being shared to you for free so I think you
> are getting the better side of the bargain here.  Asking for a bounty on a
> feature that only helps one individual is very common.............
>
>
>
>
>
> On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I said I OFFER it in case of I finish my (no money no profit but big
>> potential) experimental project.
>> I understand perfectly your mind of how to manage an open source project,
>> but don't forget
>> that there are guys like you on this emailist who don't even make any
>> decent salary, that is why
>> I use and develop open source : share -> evovle your minds -> make all
>> people happy -> money comes
>> but not                                         share -> ask money -> make
>> people weird -> money won't come
>> it's my point of view.
>>
>> Regards
>>
>> ----- Original Message -----
>>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 1:10 PM
>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> Aww, The les paul sounded nice I have one on my wishlist =D
>> http://bit.ly/bRHKlk
>>
>> *shrug* $250.00 maybe?
>>
>>
>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  OK How much ?
>>>
>>>  ----- Original Message -----
>>>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> You would need to make a patch for it or pay someone to add it.
>>>
>>>
>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>> is not compatible)
>>>>
>>>> Thanks
>>>>
>>>> F
>>>>
>>>>  ----- Original Message -----
>>>>  *From:* Madovsky <infos at madovsky.org>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>> *Subject:* vars from sip_registrations table
>>>>
>>>> in sip_registration table there are
>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>
>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>> I need to replace the host of the callee in case of if he's registered
>>>> on another FS node.
>>>>
>>>> thanks
>>>>
>>>> Franck
>>>>
>>>>
>>>> _______________________________________________
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>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
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>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>
>>> _______________________________________________
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rm at callrica.co.za  Tue Mar  9 11:08:42 2010
From: rm at callrica.co.za (Roly Maz)
Date: Tue, 9 Mar 2010 21:08:42 +0200
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
Message-ID: <000001cabfbb$f2114c90$d633e5b0$@co.za>

Ah I see... what? Please share and lead this blind man out the FS
wilderness!

I don't understand...what happens to the external profile? Do you delete it?
And how do you forward port 5060? 

...and you thought you were a newbie!

Any insight would be much appreciated...loving the journey.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 09 March 2010 08:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

Ah I see.  I will try again using the internal profile and forwarding  
port 5060.  Presumably still creating a directory entry to enable the  
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two  
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing  
>> something wrong here as I am still having difficulty, and still  
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding  
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to  
>> retain an entry in the directory /usr/local/freeeswitch/conf/ 
>> directory/ext_dns.xml in order to give freeswitch the dns name of  
>> the server as a domain for the remote softphone to register on.  I  
>> left the group name in this entry the same as inthe default entry,  
>> so that the remote phone could register on the same extension  
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN  
>> as freeswitch) phone to the remote phone, I get the message on the  
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of  
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment  
>> <param name="aggressive-nat-detection" value="true"/> in  
>> external.xml, otherwise in the case of a call initiated by the  
>> remote phone being hung up by the local phone, freeswitch sent the  
>> BYE to the private IP of the remote phone, rather than its public  
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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




From infos at madovsky.org  Tue Mar  9 11:09:06 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:09:06 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <AF70391FDBF24706A96129C21BF4C3F1@MOBILEE1705>

>This includes most of the core, and 90% of the modules you are playing with ..
do you think I really "play" with ? it's like you are a teacher, complaints about your knowledge and complaint to your students
you worked for nothing for years. 
24000 lines ? it's nothing if I compare to a project I offer fro free for NGOs  that contains 800 000.lines do you want to continue to compete
and complaint ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:45 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  ORLY,

  So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
  So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

  By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............





  On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

    I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
    I understand perfectly your mind of how to manage an open source project, but don't forget
    that there are guys like you on this emailist who don't even make any decent salary, that is why
    I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
    but not                                         share -> ask money -> make people weird -> money won't come
    it's my point of view.

    Regards
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 1:10 PM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      Aww, The les paul sounded nice I have one on my wishlist =D 
      http://bit.ly/bRHKlk

      *shrug* $250.00 maybe?



      On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

        OK How much ?
          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 11:39 AM
          Subject: Re: [Freeswitch-users] vars from sip_registrations table


          You would need to make a patch for it or pay someone to add it.



          On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

            forgot to say that I need it (since I didn't find any good examples)
            for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 09, 2010 2:59 AM
              Subject: vars from sip_registrations table


              in sip_registration table there are
                    call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

              fields, so how can I get these vars from the callee before the bridge ?
              I need to replace the host of the callee in case of if he's registered on another FS node.

              thanks

              Franck

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





          -- 
          Anthony Minessale II

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

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

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



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      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



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  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
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  pstn:+19193869900



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From sean at obscuradigital.com  Tue Mar  9 11:08:38 2010
From: sean at obscuradigital.com (Obscura)
Date: Tue, 9 Mar 2010 11:08:38 -0800
Subject: [Freeswitch-users] Outbound calls mobile
In-Reply-To: <10e701cabfb8$fa211440$ee633cc0$@com>
References: <107001cabfb3$891cd5e0$9b5681a0$@com>
	<C7BBD3C2.5FF6%sean@obscuradigital.com>
	<10e701cabfb8$fa211440$ee633cc0$@com>
Message-ID: <90B41691-BD5F-4E36-B8D6-B394889E3ADF@obscuradigital.com>

I'm using iCall carrier service gateway.  IP based authentication.  I  
know it's worked in the past but ever since I did a svn update it  
hasn't worked speifically connecting to mobile/cell phones.  Not sure  
if this answers question.

Sean
Sent from my iPhone
On Mar 9, 2010, at 10:47 AM, "Peder" <peder at networkoblivion.com> wrote:

> Is this out a sip trunk, or a PRI, or some other gateway?
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Sean Holt
> Sent: Tuesday, March 09, 2010 12:30 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Outbound calls mobile
>
>
>
>
> Basically I have sip trace turned on and I?m watching the call.  It  
> gets to the point where it is ?Trying? to esbstalish the call  
> then for no reason it times out.  No further details from the sip tr 
> ace other then call has timed out.
>
> The strange part is this doesn?t happen when calling any analog lin 
> e.
>
> If you want I can provide the sip trace.
>
> Sean
>
>
> On 3/9/10 10:08 AM, "Peder" <peder at networkoblivion.com> wrote:
>
> What are the problems?
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Sean Holt
> Sent: Tuesday, March 09, 2010 11:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Outbound calls mobile
>
> Hello, having weird issue with outbound calls to any mobile phone,  
> but not to LAN/analog lines.  Curious if anyone else has had this  
> experienced with their freeswitch setup.
>
> My setup:
> Centos 5.4 32bit
> Freeswitch 1.0.5 most recent svn trunk
>
> Thanks
> Sean
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> users
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From infos at madovsky.org  Tue Mar  9 11:11:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:11:58 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
Message-ID: <3C08E64A35614EE68E5936C80E809F9B@MOBILEE1705>

In more let me say to you that my "dozen" of emails  I sent to this emailist were not "help"
but info request because of doc lacks... So anyway, I don't care what do you think about me.
but the thing I'm sure is I don't complaint of my situation since I'm not the center of the earth...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 1:45 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  ORLY,

  So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
  So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

  By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............





  On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

    I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
    I understand perfectly your mind of how to manage an open source project, but don't forget
    that there are guys like you on this emailist who don't even make any decent salary, that is why
    I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
    but not                                         share -> ask money -> make people weird -> money won't come
    it's my point of view.

    Regards
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 09, 2010 1:10 PM
      Subject: Re: [Freeswitch-users] vars from sip_registrations table


      Aww, The les paul sounded nice I have one on my wishlist =D 
      http://bit.ly/bRHKlk

      *shrug* $250.00 maybe?



      On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

        OK How much ?
          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 09, 2010 11:39 AM
          Subject: Re: [Freeswitch-users] vars from sip_registrations table


          You would need to make a patch for it or pay someone to add it.



          On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

            forgot to say that I need it (since I didn't find any good examples)
            for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 09, 2010 2:59 AM
              Subject: vars from sip_registrations table


              in sip_registration table there are
                    call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

              fields, so how can I get these vars from the callee before the bridge ?
              I need to replace the host of the callee in case of if he's registered on another FS node.

              thanks

              Franck

            _______________________________________________
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            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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          -- 
          Anthony Minessale II

          FreeSWITCH http://www.freeswitch.org/
          ClueCon http://www.cluecon.com/
          Twitter: http://twitter.com/FreeSWITCH_wire

          AIM: anthm
          MSN:anthony_minessale at hotmail.com
          GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
          IRC: irc.freenode.net #freeswitch

          FreeSWITCH Developer Conference
          sip:888 at conference.freeswitch.org
          iax:guest at conference.freeswitch.org/888
          googletalk:conf+888 at conference.freeswitch.org
          pstn:+19193869900



----------------------------------------------------------------------


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          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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From infos at madovsky.org  Tue Mar  9 11:14:53 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:14:53 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705><191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
Message-ID: <B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>

Ok, tell me where I insulted you ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:06 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Also, when you are done insulting me, maybe try sofia_contact api command that will resolve your destination to the exact contact string that the phone registered with which is really the only bit of info
  that you would possibly need rather than asking for another expensive database lookup per call.



  On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

    ORLY,

    So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
    So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

    By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............






    On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

      I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
      I understand perfectly your mind of how to manage an open source project, but don't forget
      that there are guys like you on this emailist who don't even make any decent salary, that is why
      I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
      but not                                         share -> ask money -> make people weird -> money won't come
      it's my point of view.

      Regards
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 1:10 PM
        Subject: Re: [Freeswitch-users] vars from sip_registrations table


        Aww, The les paul sounded nice I have one on my wishlist =D 
        http://bit.ly/bRHKlk

        *shrug* $250.00 maybe?



        On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

          OK How much ?
            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 09, 2010 11:39 AM
            Subject: Re: [Freeswitch-users] vars from sip_registrations table


            You would need to make a patch for it or pay someone to add it.



            On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

              forgot to say that I need it (since I didn't find any good examples)
              for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

              Thanks

              F
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 09, 2010 2:59 AM
                Subject: vars from sip_registrations table


                in sip_registration table there are
                      call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

                fields, so how can I get these vars from the callee before the bridge ?
                I need to replace the host of the callee in case of if he's registered on another FS node.

                thanks

                Franck

              _______________________________________________
              FreeSWITCH-users mailing list
              FreeSWITCH-users at lists.freeswitch.org
              http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
              UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
              http://www.freeswitch.org





            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



--------------------------------------------------------------------


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        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



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    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



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From infos at madovsky.org  Tue Mar  9 11:16:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 9 Mar 2010 14:16:08 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705><191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com><509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705><191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com><1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705><191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
Message-ID: <49F280E71F554CFD8BDCD78E53F14A71@MOBILEE1705>

Thank you !
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 09, 2010 2:06 PM
  Subject: Re: [Freeswitch-users] vars from sip_registrations table


  Also, when you are done insulting me, maybe try sofia_contact api command that will resolve your destination to the exact contact string that the phone registered with which is really the only bit of info
  that you would possibly need rather than asking for another expensive database lookup per call.



  On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

    ORLY,

    So maybe if you *think* you know what you are talking about, you should go do some research on the average amount of code I write on a daily basis nonstop for the last several years without "ask money".  This includes most of the core, and 90% of the modules you are playing with right now including the 24 thousand lines of code in mod_sofia alone.
    So frankly, since I don't see an urgent need for your request, and I am too busy to do it, this particular feature would require a bounty to complete.  

    By the way, you have sent like 2 dozen emails in the same week asking for all kinds of help, that's also being shared to you for free so I think you are getting the better side of the bargain here.  Asking for a bounty on a feature that only helps one individual is very common.............






    On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:

      I said I OFFER it in case of I finish my (no money no profit but big potential) experimental project.
      I understand perfectly your mind of how to manage an open source project, but don't forget
      that there are guys like you on this emailist who don't even make any decent salary, that is why
      I use and develop open source : share -> evovle your minds -> make all people happy -> money comes
      but not                                         share -> ask money -> make people weird -> money won't come
      it's my point of view.

      Regards
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 09, 2010 1:10 PM
        Subject: Re: [Freeswitch-users] vars from sip_registrations table


        Aww, The les paul sounded nice I have one on my wishlist =D 
        http://bit.ly/bRHKlk

        *shrug* $250.00 maybe?



        On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

          OK How much ?
            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 09, 2010 11:39 AM
            Subject: Re: [Freeswitch-users] vars from sip_registrations table


            You would need to make a patch for it or pay someone to add it.



            On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:

              forgot to say that I need it (since I didn't find any good examples)
              for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible)

              Thanks

              F
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 09, 2010 2:59 AM
                Subject: vars from sip_registrations table


                in sip_registration table there are
                      call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname  

                fields, so how can I get these vars from the callee before the bridge ?
                I need to replace the host of the callee in case of if he's registered on another FS node.

                thanks

                Franck

              _______________________________________________
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            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



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        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



------------------------------------------------------------------------


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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



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From msc at freeswitch.org  Tue Mar  9 11:18:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 11:18:39 -0800
Subject: [Freeswitch-users] call forward
In-Reply-To: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>
References: <883b62721003090940h25c3a853pdda2c281555caeea@mail.gmail.com>
Message-ID: <87f2f3b91003091118o1191c519wb2972c1c8f9fc087@mail.gmail.com>

On Tue, Mar 9, 2010 at 9:40 AM, chandrasekhar reddy <sekharreddypk at gmail.com
> wrote:

> Hi,
>
> i am trying to forward call from freeswitch to cisco router, call is
> estabilished but after 30sec call got ended, any idea
>
>

Sounds like a timeout issue. Make sure that there's no naughty NAT router in
between your devices. You could also get a pcap of the traffic between the
two devices and see where the dialog goes awry.
-MC
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From anthony.minessale at gmail.com  Tue Mar  9 11:41:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Mar 2010 13:41:30 -0600
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<191c3a031003091045w11a564d0r9757a2ae44a4542b@mail.gmail.com>
	<191c3a031003091106g703866b2r863a57e70c235c28@mail.gmail.com>
	<B463A3632FEC4E5FA540082AE8224529@MOBILEE1705>
Message-ID: <191c3a031003091141i32599500o338f86d225bd3cb9@mail.gmail.com>

Well every email you sent since i suggested you file a bounty request for
your feature is more or less carrying an insulting tone.  Go back and read
it.  You are insulting everyone on this list with your attempts to downplay
the effort the community has made to create this project.

An on the subject of being able to read, I said "mod_sofia" (1 module in FS)
is 24000 lines.
This is not a contest about how many lines of code but since you feel the
need to continue:

http://fisheye.freeswitch.org/committer/FreeSWITCH/anthm

FreeSWITCH has over 3 million lines of code in SVN.

You can clearly see I have personally committed 600,000 lines of code in 4
years and started this project from nothing.
The only point of this information is that almost all of this code was
written completely for free.
This means you are completely wrong with your previous statements that
everything is for money.
The actual amount of code written and time spent is not even relevant apart
from that.

I only ask for money for things I don't want to do and someone else
wants me to do it.  In fact, even if you paid me, I would not add your
feature so forget it.
The more I thought about it the more I decided it's a waste of resources on
the application.

If our work is insignificant to you, I don't really care.  Feel free to
write your own code instead.
Keep in mind that if you continue to abuse and flame our mailing list, you
will be moderated.

If you can be civil and stop the flaming you are welcome to stay.

I recommend an apology to the community to start with.
Also, try to consolidate your thoughts into 1 single email instead of
sending 5 in a row on the same thread.


On Tue, Mar 9, 2010 at 1:14 PM, Madovsky <infos at madovsky.org> wrote:

>  Ok, tell me where I insulted you ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 2:06 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Also, when you are done insulting me, maybe try sofia_contact api command
> that will resolve your destination to the exact contact string that the
> phone registered with which is really the only bit of info
> that you would possibly need rather than asking for another expensive
> database lookup per call.
>
>
> On Tue, Mar 9, 2010 at 12:45 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> ORLY,
>>
>> So maybe if you *think* you know what you are talking about, you should go
>> do some research on the average amount of code I write on a daily basis
>> nonstop for the last several years without "ask money".  This includes most
>> of the core, and 90% of the modules you are playing with right now including
>> the 24 thousand lines of code in mod_sofia alone.
>> So frankly, since I don't see an urgent need for your request, and I am
>> too busy to do it, this particular feature would require a bounty to
>> complete.
>>
>> By the way, you have sent like 2 dozen emails in the same week asking for
>> all kinds of help, that's also being shared to you for free so I think you
>> are getting the better side of the bargain here.  Asking for a bounty on a
>> feature that only helps one individual is very common.............
>>
>>
>>
>>
>>
>> On Tue, Mar 9, 2010 at 12:25 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  I said I OFFER it in case of I finish my (no money no profit but big
>>> potential) experimental project.
>>> I understand perfectly your mind of how to manage an open source project,
>>> but don't forget
>>> that there are guys like you on this emailist who don't even make any
>>> decent salary, that is why
>>> I use and develop open source : share -> evovle your minds -> make all
>>> people happy -> money comes
>>> but not                                         share -> ask money ->
>>> make people weird -> money won't come
>>> it's my point of view.
>>>
>>> Regards
>>>
>>>  ----- Original Message -----
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 1:10 PM
>>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> Aww, The les paul sounded nice I have one on my wishlist =D
>>> http://bit.ly/bRHKlk
>>>
>>> *shrug* $250.00 maybe?
>>>
>>>
>>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  OK How much ?
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>>
>>>> You would need to make a patch for it or pay someone to add it.
>>>>
>>>>
>>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>>> is not compatible)
>>>>>
>>>>> Thanks
>>>>>
>>>>> F
>>>>>
>>>>>  ----- Original Message -----
>>>>> *From:* Madovsky <infos at madovsky.org>
>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>>> *Subject:* vars from sip_registrations table
>>>>>
>>>>> in sip_registration table there are
>>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>>
>>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>>> I need to replace the host of the callee in case of if he's registered
>>>>> on another FS node.
>>>>>
>>>>> thanks
>>>>>
>>>>> Franck
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> ------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Tue Mar  9 12:43:37 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 9 Mar 2010 21:43:37 +0100
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
Message-ID: <4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>

This is hypochisy  in action.

99% of people asking questions in this ml are using FreeSWITCH for
commercial use, I for one.
So if the primary author of software that gets you paid asks for small
bounty, I'd shut up or pay up.


Ognjen

On Tue, Mar 9, 2010 at 7:25 PM, Madovsky <infos at madovsky.org> wrote:

>  I said I OFFER it in case of I finish my (no money no profit but big
> potential) experimental project.
> I understand perfectly your mind of how to manage an open source project,
> but don't forget
> that there are guys like you on this emailist who don't even make any
> decent salary, that is why
> I use and develop open source : share -> evovle your minds -> make all
> people happy -> money comes
> but not                                         share -> ask money -> make
> people weird -> money won't come
> it's my point of view.
>
> Regards
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 09, 2010 1:10 PM
> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>
> Aww, The les paul sounded nice I have one on my wishlist =D
> http://bit.ly/bRHKlk
>
> *shrug* $250.00 maybe?
>
>
> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  OK How much ?
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>  *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> You would need to make a patch for it or pay someone to add it.
>>
>>
>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  forgot to say that I need it (since I didn't find any good examples)
>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is
>>> not compatible)
>>>
>>> Thanks
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>> *Subject:* vars from sip_registrations table
>>>
>>> in sip_registration table there are
>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>
>>> fields, so how can I get these vars from the callee before the bridge ?
>>> I need to replace the host of the callee in case of if he's registered on
>>> another FS node.
>>>
>>> thanks
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.sobkow at marketelsystems.com  Tue Mar  9 13:57:01 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 15:57:01 -0600
Subject: [Freeswitch-users] IVR and Erlang
In-Reply-To: <4B703921.4080909@marketelsystems.com>
References: <4B70288F.6040003@marketelsystems.com>
	<4B703921.4080909@marketelsystems.com>
Message-ID: <4B96C42D.5020609@marketelsystems.com>

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From m.sobkow at marketelsystems.com  Tue Mar  9 14:46:16 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 09 Mar 2010 16:46:16 -0600
Subject: [Freeswitch-users] Call keeps hanging up
In-Reply-To: <20100304234050.GL1751@hijacked.us>
References: <4B8FFDA7.50708@marketelsystems.com>	<20100304184755.GK1751@hijacked.us>	<4B9035B4.6030901@marketelsystems.com>	<87f2f3b91003041500p18f4b523mbf545c30a0612813@mail.gmail.com>
	<20100304234050.GL1751@hijacked.us>
Message-ID: <4B96CFB8.6060803@marketelsystems.com>

I spoke with "the boss" today, and he's hoping to do a donation at month 
end if our accountant can confirm that it can be taken as a tax writeoff 
(I suggested it's an alternative way of paying for software, so it 
should be write-offable.)

Andrew Thompson wrote:
> On Thu, Mar 04, 2010 at 03:00:12PM -0800, Michael Collins wrote:
>
>   
>>> "Another satisfied customer!" :D Now it's time to throw some of that
>>>       
>> Canadian colored money at Tony's paypal account. ;)
>>     
>
> Hey! I wrote the erlang module and provided the solution to the problem
> ;)
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From craig at overthewire.com.au  Tue Mar  9 15:25:31 2010
From: craig at overthewire.com.au (Craig Askings)
Date: Wed, 10 Mar 2010 09:25:31 +1000
Subject: [Freeswitch-users] freeswitch to cisco router
In-Reply-To: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
References: <f8af5741003090547i4e27bf26hb2f6f5b10fa95f14@mail.gmail.com>
Message-ID: <8cc991dd1003091525t2600adf1m3c45dbf5ef62e7e6@mail.gmail.com>

Hi Swapna,

Are you trying to register the Cisco to the Freeswitch server? or the other
way around. If the former check out the following documentation.

http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter1.html#wp1182326

sip-ua
  authentication username [INSERT_EXTENTION] password [INSERT_PASSWORD]
  registrar ipv4:[INERST_NAME_OF_FS_BOX]:5060 expires 3600
  sip-server ipv4:[INERST_NAME_OF_FS_BOX]:5060

Regards,
<http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter1.html#wp1182326>
Craig.

On 9 March 2010 23:47, srinivasula reddy <srinivas.ksvreddy at gmail.com>wrote:

> Hi,
>
> i have tried to configure gateway from my freeswitch, with the following
> credentials proxy(ip of cisco) and username,password.
> and when i start my freeswitch registration reqest is going to cisco router
> and again it requesting to same registration reqeust to my sipserver, the
> register packet from cisco router to my freeswitch is comming like this
>
> To: sip:.*@sipserverip:5060
>
> so there there is no user present with the name of .* it is rejecting the
> registration request, any idea
>
> Thanks
> Swapna
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Craig Askings

Network Engineer | Over the Wire Pty Ltd
craig at overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
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From lloyd.aloysius at gmail.com  Tue Mar  9 16:57:33 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 9 Mar 2010 19:57:33 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
Message-ID: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>

Hi All,

I am trying to setup a simple IVR with two options , timeout action and
Invalid digits pressed. here is the partial IVR Menu.


<entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
default"/>
<entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
default"/>


How to Catch Timeout and Invalid Digits Actions from the IVR menu?

Thanks you

Thanks,
Lloyd
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From frank at carmickle.com  Tue Mar  9 17:17:40 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 9 Mar 2010 20:17:40 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
In-Reply-To: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
References: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
Message-ID: <20100310011738.GS18427@base.carmickle.com>

Hello

On Tue, Mar 09, Aloysius Lloyd wrote:
> Hi All,
> 
> I am trying to setup a simple IVR with two options , timeout action and
> Invalid digits pressed. here is the partial IVR Menu.
> 
> 
> <entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
> default"/>
> <entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
> default"/>

Add something like.

                timeout="3000"
                inter-digit-timeout="2000"
                max-failures="3"
                max-timeouts="3"
 
Then in the dialplan after you've called the ivr tranfer to another exten.  Be careful to make sure that hangup after bridge gets set so that you don't ring the exten when someone who has been called from the ivr hangs up.

HTH
--FC



From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:15:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:15:29 -0500
Subject: [Freeswitch-users] actition after a set time during call
In-Reply-To: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
References: <BLU110-DS12F13B3418ABF2053C86AAA73F0@phx.gbl>
Message-ID: <BF002F6F-D2B4-496F-A666-37746D126044@jerris.com>

You can use execute_on_answer in combination with sched_api or something similar to acomplish this.

Mike

On Feb 26, 2010, at 1:53 PM, Todd wrote:

> Hey List-
>  
> I want to have nibblebill pause after a certain time during a call?  I was wondering what the best way to put this into the dialplan is?  Still kind of new to this?. 
>  
>  <action application="nibblebill" data="pause"/>
>  
> Is the action I want to implement 2 minutes into a call? what is the best way to do this?  I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml  Thanks
>  
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From mike at jerris.com  Tue Mar  9 16:20:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:20:02 -0500
Subject: [Freeswitch-users] How to tie context to a gateway?
In-Reply-To: <20100216141634.75B3511FC6@mail.nstel.ru>
References: <20100216141634.75B3511FC6@mail.nstel.ru>
Message-ID: <7BA30ABD-5886-42AF-B22F-FBC4D13FDCE0@jerris.com>

You can use conditions and transfer, the issue is, the inbound call has to be matched on the gateway, which means the other endpoint must actually send the request to our registered contact.  Many do not do this.  Its probably easier to just use multiple profiles on multiple ports.

Mike


On Feb 16, 2010, at 9:16 AM, Nikolay Kondratyev wrote:

> Hi all,
> I have several gateways in the external profile.
> Let?s say GW1 and GW2. I?d like to process calls from the GW1 in the context C1 and calls from GW2 in the context C2.
> Parameter ?context?, as far as I understand works for the whole profile, not for individual gateways in the profile.
> How do send calls from GW1 into context C1?
> What will be a good practice to do that?
> Thanks in advance,
> Nikolay.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Mar  9 16:21:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:21:32 -0500
Subject: [Freeswitch-users] call routing from freeswitch based on INVITE
In-Reply-To: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>
References: <f8af5741002282339l52fe9ad7q44a7e4c65f82cd95@mail.gmail.com>
Message-ID: <EEB17718-A279-4E5F-BBF1-5C60765A43A3@jerris.com>

if you look at the vars from the info app for that inbound call, you will see all the relevant parts of the sip packet there to use in a condition.

Mike

On Mar 1, 2010, at 2:39 AM, srinivasula reddy wrote:

>  
>  
> I have two sipservers like server1 and server2,  if sever1 receives invite packet like
>  
> INVITE From:   1000 at server1.domain.com
>            To:   1002 at server2.railvoice.com.
>  
> how can i route the invite packet to server2 from server1, 

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From mike at jerris.com  Tue Mar  9 16:25:55 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 9 Mar 2010 19:25:55 -0500
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
	<alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
Message-ID: <034FE08F-7E01-45DD-8758-61FEA5AEEAF1@jerris.com>

Could you please contribute a patch or at least report this issue to http;//jira.frreeswitch,org

On Mar 1, 2010, at 10:29 PM, Wellie Chao wrote:

> Terrific. "show calls as xml" works great and just what I needed.
> 
> P.S. "show calls as csv" still has the comma bug. Not a big deal since I 
> can get it as XML. CSV needs quotes around field values that contain a 
> comma. Pretty easy to fix. Few lines of C (strchr for "," and if found, 
> put quotes around field value).
> 




From pjintheusa at gmail.com  Tue Mar  9 18:15:15 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Mar 2010 21:15:15 -0500
Subject: [Freeswitch-users] How to originate a new call from mod_managed?
In-Reply-To: <9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>
References: <9394d42f1003062102p3d97f86haa81c009f19be817@mail.gmail.com>
	<367751821003081204t7b18d8ebkb11bbd27828e01f6@mail.gmail.com>
	<9394d42f1003081830t3257b63av1de166b2ae8038e7@mail.gmail.com>
Message-ID: <367751821003091815q72cce2bey6399b90c657a61f8@mail.gmail.com>

After originate I call a method:

Uuid = fsApi.ExecuteString("create_uuid");
apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/g1/{2}
'&managed(MyFSProjects.TestCallAnswered)'", Uuid, OutgoingCallerID,
NumberToDial));

My "TestCallAnswered" has reference to the session:

public void Run(AppContext context)
        {
            BLegSession = context.Session;

....

With the UUID you can also call other API functions:

apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}",
Aleg.Uuid, BLegSession.GetUuid()));
apiResult = fsApi.Execute("uuid_media", "off " + call.Uuid);
apiResult = fsApi.ExecuteString(string.Format("uuid_kill {0}", call.Uuid));

Hope that helps


On Mon, Mar 8, 2010 at 9:30 PM, Chris <freeswitch at gilligan.id.au> wrote:

> That decently woks to originate the call but passing that UUID into a new
> ManagedSession does not seem to like to the session created using that UUID.
>  Once i create a session used the API you suggested how would i then access
> the session in the Managed Code?
>
> Chris
>
> On Tue, Mar 9, 2010 at 7:04 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> You could use the api:
>>
>>  FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api();
>> string apiResult = string.Empty;
>> string Uuid;
>> string NumberToDial = "3475558308";
>> string OutgoingCallerID = "2155556666";
>>
>> Uuid = fsApi.ExecuteString("create_uuid");
>>
>> apiResult = fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}",
>> Uuid, OutgoingCallerID, NumberToDial));
>>
>>
>>
>>
>>
>>
>> On Sun, Mar 7, 2010 at 12:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>
>>> Hi,
>>> I am trying to create a mod_managed API application that takes 2 phone
>>> numbers as params.  These numbers should then be used to make 2 calls and to
>>> bridge them. The issue i am having is working out how to place the first
>>> call leg from in mod_managed.  All the methods i have found are for the
>>> second leg and require you to pass in the first call leg.
>>>
>>> I know there are easier way to do this out of the managed code but i want
>>> it in the managed code as this is just a proof of concept to prove we can do
>>> a callback like system via managed code to be hooked into other external
>>> systems.
>>> In wiki http://wiki.freeswitch.org/wiki/Session there is this example
>>>
>>> s = new Session("{ignore_early_media=true}sofia/default/foo at bar.com");
>>> while (s.ready()) {
>>> // The call has been answered
>>> }
>>>
>>> This seems to be exactly what i am looking for but seems to be missing
>>> from mod_managed. I am hoping someone can tell me how to do something
>>> similar in mod_managed since even if it is not part of the managed code
>>> mod_managed is meant to have the native access as well so i would assume it
>>> would be possible.
>>>
>>> Thanks in advance
>>>
>>> Chris
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wiltingtree at gmail.com  Tue Mar  9 18:26:06 2010
From: wiltingtree at gmail.com (Adam Wilt)
Date: Tue, 9 Mar 2010 21:26:06 -0500
Subject: [Freeswitch-users] vars from sip_registrations table
In-Reply-To: <4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>
References: <806942C2ED024A408C98C207AEC3FA70@MOBILEE1705>
	<191c3a031003090839t41d16138o9e0e49683cc5ff9e@mail.gmail.com>
	<509EA502A6AE46E88BE8A205B74834B7@MOBILEE1705>
	<191c3a031003091010u6d0a9ba9r9d3c0c753396489c@mail.gmail.com>
	<1F4E428CA9754FE49A8D16442657CE4C@MOBILEE1705>
	<4468a6771003091243q1678cfe8o38ea92bc6b05aeca@mail.gmail.com>
Message-ID: <eb7797d71003091826j3fb9bb73k578aee8eed17a26e@mail.gmail.com>

So Anthony, don't take this the wrong way, and I'm sorry if I'm asking a
personal question, but if you've been writing this application for free for
the past 4 years, how do you support yourself?
Do you have a day job as well?


On Tue, Mar 9, 2010 at 3:43 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> This is hypochisy  in action.
>
> 99% of people asking questions in this ml are using FreeSWITCH for
> commercial use, I for one.
> So if the primary author of software that gets you paid asks for small
> bounty, I'd shut up or pay up.
>
>
> Ognjen
>
>
> On Tue, Mar 9, 2010 at 7:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I said I OFFER it in case of I finish my (no money no profit but big
>> potential) experimental project.
>> I understand perfectly your mind of how to manage an open source project,
>> but don't forget
>> that there are guys like you on this emailist who don't even make any
>> decent salary, that is why
>> I use and develop open source : share -> evovle your minds -> make all
>> people happy -> money comes
>> but not                                         share -> ask money -> make
>> people weird -> money won't come
>> it's my point of view.
>>
>> Regards
>>
>> ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 09, 2010 1:10 PM
>> *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>
>> Aww, The les paul sounded nice I have one on my wishlist =D
>> http://bit.ly/bRHKlk
>>
>> *shrug* $250.00 maybe?
>>
>>
>> On Tue, Mar 9, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  OK How much ?
>>>
>>>  ----- Original Message -----
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>>  *To:* freeswitch-users at lists.freeswitch.org
>>>  *Sent:* Tuesday, March 09, 2010 11:39 AM
>>>   *Subject:* Re: [Freeswitch-users] vars from sip_registrations table
>>>
>>> You would need to make a patch for it or pay someone to add it.
>>>
>>>
>>> On Tue, Mar 9, 2010 at 2:04 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  forgot to say that I need it (since I didn't find any good examples)
>>>> for a 2 FS clustered without SRV NAPTR in DNS (the client I should use
>>>> is not compatible)
>>>>
>>>> Thanks
>>>>
>>>> F
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Madovsky <infos at madovsky.org>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>   *Sent:* Tuesday, March 09, 2010 2:59 AM
>>>> *Subject:* vars from sip_registrations table
>>>>
>>>> in sip_registration table there are
>>>>   call_id<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=1&sortdir=asc&strings=collapsed&page=1>
>>>> sip_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=2&sortdir=asc&strings=collapsed&page=1>
>>>> sip_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=3&sortdir=asc&strings=collapsed&page=1>
>>>> presence_hosts<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=4&sortdir=asc&strings=collapsed&page=1>
>>>> contact<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=5&sortdir=asc&strings=collapsed&page=1>
>>>> status<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=6&sortdir=asc&strings=collapsed&page=1>
>>>> rpid<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=7&sortdir=asc&strings=collapsed&page=1>
>>>> expires<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=8&sortdir=asc&strings=collapsed&page=1>
>>>> user_agent<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=9&sortdir=asc&strings=collapsed&page=1>
>>>> server_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=10&sortdir=asc&strings=collapsed&page=1>
>>>> server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=11&sortdir=asc&strings=collapsed&page=1>
>>>> profile_name<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=12&sortdir=asc&strings=collapsed&page=1>
>>>> hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=13&sortdir=asc&strings=collapsed&page=1>
>>>> network_ip<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=14&sortdir=asc&strings=collapsed&page=1>
>>>> network_port<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=15&sortdir=asc&strings=collapsed&page=1>
>>>> sip_username<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=16&sortdir=asc&strings=collapsed&page=1>
>>>> sip_realm<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=17&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_user<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=18&sortdir=asc&strings=collapsed&page=1>
>>>> mwi_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=19&sortdir=asc&strings=collapsed&page=1>
>>>> orig_server_host<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=20&sortdir=asc&strings=collapsed&page=1>
>>>> orig_hostname<https://secure.live-school.net/phpPgAdmin4/display.php?server=10.0.0.1%3A5432&database=freeswitch&schema=public&table=sip_registrations&subject=table&return_url=tblproperties.php%3Fserver%3D10.0.0.1%253A5432%26amp%3Bdatabase%3Dfreeswitch%26amp%3Bschema%3Dpublic%26table%3Dsip_registrations&return_desc=Back&sortkey=21&sortdir=asc&strings=collapsed&page=1>
>>>>
>>>> fields, so how can I get these vars from the callee before the bridge ?
>>>> I need to replace the host of the callee in case of if he's registered
>>>> on another FS node.
>>>>
>>>> thanks
>>>>
>>>> Franck
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Mar  9 18:31:17 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 9 Mar 2010 21:31:17 -0500
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
Message-ID: <367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>

One way might be to create an event loop and listen for DTMF

public bool Load()
        {
            ThreadPool.QueueUserWorkItem((o) =>
            {
                Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
                EventConsumer con = new EventConsumer("DTMF", "");
                while (true)
                {
                    Event ev = con.pop(1);
                    if (ev != null)
                    {
                                  ... do what ever needs to be done
                    }

                }
            });

See http://wiki.freeswitch.org/wiki/Event_List#DTMF

I am guessing the DTMF is "DTMF" so play around with that if it does not
work.

There are probably better ways.



On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:

> What i need to do is listen on a session before, during and after the
> session is bridged with another call. I know built into FS is the option to
> listen for *x where x is 0-9 but i really need more than one digit.
>
> I traced the method that listens for the single digit to the
> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
> switch_ivr_async.c
>
> mod_managed also has the following in the native class freeswitch.
> public static switch_status_t
> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
> recv_dtmf)
>
> The problem is i can't work out how to use it.  I am fine the the first
> param for the session but have no idea what i need for
> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
> recv_dtmf param.
>
> if anyone can point me in the right direction or knows a better way i would
> be grateful.
>
> Chris
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Mar  9 19:49:47 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 10 Mar 2010 11:49:47 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
Message-ID: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>

I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.

One running trunk. and back ported to another server running rev14696.
Sounds great.


2010/3/9 jay binks <jaybinks at gmail.com>:
> this is what I love about Freeswitch, and the FS development team.
> Well done Brian ( and the rest of the team ).
> Jay
>
>
> On Tue, Mar 9, 2010 at 11:07 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> We now have mod_silk in tree it has only been tested on Linux so far. ?The
>> silk library itself was released in the IETF draft only an hour and twenty
>> minutes ago. ?I wrote a perl script to extract the source from the draft.
>> ?Checked in the lib and the codec module which I had written to the binary
>> lib I had a few months ago.
>>
>> It has more work to do... but its there if anyone wants to chip in and
>> libtoolize the library, revamp the build system and assist in testing the
>> codec on multiple platforms.
>>
>> Thanks,
>> Brian
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Tue Mar  9 20:09:11 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Mar 2010 22:09:11 -0600
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
Message-ID: <5FA53946-544F-499B-9690-25E73FB5FA19@freeswitch.org>

Thanks.

/b

On Mar 9, 2010, at 9:49 PM, Seven Du wrote:

> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
> 
> One running trunk. and back ported to another server running rev14696.
> Sounds great.




From lloyd.aloysius at gmail.com  Tue Mar  9 20:59:14 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 9 Mar 2010 23:59:14 -0500
Subject: [Freeswitch-users] IVR Menu - Timeout and Invalid Options
In-Reply-To: <20100310011738.GS18427@base.carmickle.com>
References: <8a19bf2e1003091657l578a007ej703b2b58e7e962d6@mail.gmail.com>
	<20100310011738.GS18427@base.carmickle.com>
Message-ID: <8a19bf2e1003092059o6895c8b3yceb8590d4f4be525@mail.gmail.com>

Frank ,

Thank you for your help.

Lloyd

On Tue, Mar 9, 2010 at 8:17 PM, Frank Carmickle <frank at carmickle.com> wrote:

> Hello
>
> On Tue, Mar 09, Aloysius Lloyd wrote:
> > Hi All,
> >
> > I am trying to setup a simple IVR with two options , timeout action and
> > Invalid digits pressed. here is the partial IVR Menu.
> >
> >
> > <entry action="menu-exec-app" digits="1" param="execute_extension 201 XML
> > default"/>
> > <entry action="menu-exec-app" digits="2" param="execute_extension 202 XML
> > default"/>
>
> Add something like.
>
>                timeout="3000"
>                inter-digit-timeout="2000"
>                max-failures="3"
>                max-timeouts="3"
>
> Then in the dialplan after you've called the ivr tranfer to another exten.
>  Be careful to make sure that hangup after bridge gets set so that you don't
> ring the exten when someone who has been called from the ivr hangs up.
>
> HTH
> --FC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dome at tel.co.th  Tue Mar  9 21:10:23 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 10 Mar 2010 12:10:23 +0700
Subject: [Freeswitch-users] FS no frok mode
Message-ID: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>

Dear All,
               How about performance when start FS with -nf  (no
fork). I ask because i want to keep SQL connect (cash db) in  1
connection. but i want to know other effect like a performance ,
memory usage .. etc


Best Regrads.

Dome C.



From jason at jasonjgw.net  Tue Mar  9 21:13:44 2010
From: jason at jasonjgw.net (Jason White)
Date: Wed, 10 Mar 2010 16:13:44 +1100
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
Message-ID: <20100310051344.GA19653@jdc.jasonjgw.net>

Seven Du <dujinfang at gmail.com> wrote:
> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
> 
> One running trunk. and back ported to another server running rev14696.
> Sounds great.

Do you have a comparison with my current favourite, CELT at 48000? CELT at 32000
also gives high quality, in my opinion.

I plan to test the SILK module at some point, possibly later this week.




From will.traenkle at yahoo.com  Tue Mar  9 21:23:07 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 21:23:07 -0800 (PST)
Subject: [Freeswitch-users] hunt group - play music when trying external
	numbers
In-Reply-To: <20100308184025.GL18427@base.carmickle.com>
References: <756904.84769.qm@web57613.mail.re1.yahoo.com>
	<20100308184025.GL18427@base.carmickle.com>
Message-ID: <51379.68073.qm@web57602.mail.re1.yahoo.com>

Frank,

Thanks for the tip.

I took your advice and added the following line to my huntgroup script:

session.execute("set","ignore_early_media=true");

and it works!

Thanks again,

-Will




________________________________
From: Frank Carmickle <frank at carmickle.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Mon, March 8, 2010 10:40:25 AM
Subject: Re: [Freeswitch-users] hunt group - play music when trying external numbers

Hello

On Mon, Mar 08, William Traenkle wrote:
> I am new to freeSWITCH and this mailing list and I appreciate your support in advance.
> 
> My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried.
> 
> The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried.  

Make sure that your bridge statement includes {ignore_early_media=true}

--FC

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      
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From dujinfang at gmail.com  Tue Mar  9 21:37:57 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 10 Mar 2010 13:37:57 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <20100310051344.GA19653@jdc.jasonjgw.net>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
Message-ID: <23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>

Unfortunately no.

I only tested 8000 because at least one leg is PSTN or SIP UA which
doesn't support SILK and CELT in our scenario. We are bridging UAs
between US and China so robustness is more concerned than HZ.

I just *feel* it's ok because I only tested with my ear without
considerate packet lost or jitter.

2010/3/10 Jason White <jason at jasonjgw.net>:
> Seven Du <dujinfang at gmail.com> wrote:
>> I tested on servers running Ubuntu 8.04 64bit and Ubuntu 7.10 32bit.
>>
>> One running trunk. and back ported to another server running rev14696.
>> Sounds great.
>
> Do you have a comparison with my current favourite, CELT at 48000? CELT at 32000
> also gives high quality, in my opinion.
>
> I plan to test the SILK module at some point, possibly later this week.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From will.traenkle at yahoo.com  Tue Mar  9 21:55:59 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 9 Mar 2010 21:55:59 -0800 (PST)
Subject: [Freeswitch-users] IVR to hunt group to external destination fails
	// direct to hunt group to external desitnation succeeds
Message-ID: <153857.43476.qm@web57613.mail.re1.yahoo.com>

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.

However, if you access the hunt group directly without going through the main IVR, all works perfectly.  

Any ideas where I should look and what line of code I might be missing?

Thanks,

-Will



      
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From msc at freeswitch.org  Tue Mar  9 22:50:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 22:50:59 -0800
Subject: [Freeswitch-users] FreeSWITCH Conference Call Agenda
Message-ID: <87f2f3b91003092250r6c32860ctde57e87563975f10@mail.gmail.com>

Hello all!

The FreeSWITCH weekly conference call agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_10

Please feel free to add your questions and agenda items. Also, remember that
this week Mathieu Rene is going to be spending a lot of time discussing
mod_sofia and how it works. It should be a very informative discussion! By
all means join us at 1700 GMT on Wednesday.

-Michael
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From msc at freeswitch.org  Tue Mar  9 23:00:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 9 Mar 2010 23:00:53 -0800
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <153857.43476.qm@web57613.mail.re1.yahoo.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>

On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com>wrote:

> When a hunt group is accessed from the main IVR, the hunt group dials an
> internal extension via transfer correctly but when trying to dial an
> external number such a cell phone via sip uri it is unable to connect or
> ring the phone but it waits the correct timeout period before it goes to
> voicemail.
>
> However, if you access the hunt group directly without going through the
> main IVR, all works perfectly.
>
> Any ideas where I should look and what line of code I might be missing?
>
>
Start by capturing the fs_cli debug output for the IVR call. Look through
the debug lines for clues as to what is happening with your call. The
symptom "unable to connect" has many possible causes, so try to locate the
debug lines leading up to the attempt to bridge the SIP URI. If you want
community help then pastebin the full debug log plus the relevant XML: the
dialplan extension and the IVR definition.
-MC
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From lloyd.aloysius at sunteltech.ca  Tue Mar  9 23:52:36 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 02:52:36 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly one device ring
Message-ID: <8a19bf2e1003092352r48acb809kb4bcd2c05d49eec1@mail.gmail.com>

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From lloyd.aloysius at gmail.com  Tue Mar  9 23:53:23 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 02:53:23 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
Message-ID: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From will.traenkle at yahoo.com  Wed Mar 10 00:39:41 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Wed, 10 Mar 2010 00:39:41 -0800 (PST)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
Message-ID: <474069.15451.qm@web57605.mail.re1.yahoo.com>

Michael,

Per your request:

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab 

If you require additional information or have any other questions, please let me know.

Thanks,

-Will



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:00:53 PM
Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds




On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>
>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>
>Any ideas where I should look and what line of code I might be missing?
>
>

Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
-MC


      
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From will.traenkle at yahoo.com  Wed Mar 10 00:45:40 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Wed, 10 Mar 2010 00:45:40 -0800 (PST)
Subject: [Freeswitch-users] auto attendant -> hunt group -> external
	number does not ring
In-Reply-To: <87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>
References: <302537.41702.qm@web57609.mail.re1.yahoo.com>
	<87f2f3b91003091101s2138078aj6a564ff31cd4b3cd@mail.gmail.com>
Message-ID: <533929.27849.qm@web57613.mail.re1.yahoo.com>

Michael,

There are two posts for this issue.  I just emailed the bottom links a few minutes ago, but reading your response prompted me to send one more link, which is the default.xml dialplan:

5) Dialplan default.xml: http://pastebin.com/iAbNSW9E 

Per
your request:

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab 
 
If you require additional information or have any other
questions, please let me know.

Thanks,

-Will



________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:01:31 AM
Subject: Re: [Freeswitch-users] auto attendant -> hunt group -> external number does not ring




On Mon, Mar 8, 2010 at 2:23 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

>
>FS Community,
>
>
>I created an extension
>704
> 
>I created a
>hunt group 7004 that first rings 704 then rings my cell phone 16194548924.
> 
>I created an
>auto attendant 5002 with an option of 704 that goes to hunt group 7004.
> 
>1)    
>What does
>work: When I call the hunt group directly by dialing 7004 internally, the hunt
>group first rings 704 then rings my cell phone successfully.  This is perfect.
> 
>2)    
>What does not
>work: When I call the main auto attendant from either internally by dialing 5002
>or externally by dialing my main number, and dial 704, my extension rings and
>after not answering my cell phone does not ring.
> 
>What I am expecting
>in #2 is for my extension 704 to ring and if I do not pick up then my cell
>phone will ring just like it does in #1.
> 
>My knowledge
>is limited in this area and if you could point me in the right direction, that
>would be great.  I a have tried
>everything but with no luck.
>
>
Pastebin the necessary debug info. Check http://wiki.freeswitch.org/wiki/Reporting_Bugs for specifics on how to do collect and report all the data to pastebin. It would be good to see the dialplan and the debug log for each call. 

When you've collected all the data please reply here with the pastebin URL.
-MC


      
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From tculjaga at gmail.com  Wed Mar 10 01:42:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 10:42:49 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
Message-ID: <65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>

i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...

what is this related to ?

T.

On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:

> Hi Anthony,
>
> Many, many apologies to take so long to reply - I failed to notice your
> response.  Sorry !
>
> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
> then does not give a warning.
>
> Best wishes
> Bruce
>
>
> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>
>> we are still wary about 5.3 due to bugs reported in libc.
>> What number does it say it detected for the gap?
>>
>>
>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>
>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>> ....
>>>
>>>
>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>
>>>> Hi,
>>>>
>>>> I wonder if anyone would be able to advise please:
>>>>
>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when
>>>> I start FreeSWITCH that
>>>>
>>>> "Abnormally large timer gap detected"
>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>> experience audio problems".
>>>>
>>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>>> to Centos5.4 was a yum update on the 5.3 build.
>>>>
>>>> I guess the warning comes from here:
>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>
>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>> optiplex p4's I use for testing.
>>>>
>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>> use such crummy hardware?!
>>>>
>>>> Many thanks in advance
>>>> Bruce
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed Mar 10 01:51:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:51:18 -0500
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
Message-ID: <5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>

-nf has nothing to do with the performance, it has to do if the command blocks or not when you run it.  I am unsure what you mean about sql in this context, it is totally unrelated. What is a cash db? 

Mike


On Mar 10, 2010, at 12:10 AM, Dome Charoenyost wrote:

> Dear All,
>               How about performance when start FS with -nf  (no
> fork). I ask because i want to keep SQL connect (cash db) in  1
> connection. but i want to know other effect like a performance ,
> memory usage .. etc
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From mike at jerris.com  Wed Mar 10 01:54:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:54:53 -0500
Subject: [Freeswitch-users] Incorrect nonce
In-Reply-To: <d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>
References: <d021d3be1003020702g3a773ce0l132f0cc4b8655fdb@mail.gmail.com>
	<4B8E6902.90106@fx-services.com>
	<d021d3be1003032301i58bf53eby9541e833ccef41a@mail.gmail.com>
Message-ID: <0E77A138-8D81-4A6D-BDB6-D94F25CD7B57@jerris.com>

I am not sure on this, if we are not using the same one here, I suspect that it is down in the sofia lib, and will require much more digging.  Try turing on ALL the sofia debug in the lib and see if any of the auth debug jumps out as to what might cause this.

Mike

On Mar 4, 2010, at 2:01 AM, Jonas Gauffin wrote:

> You are correct, I'm using DNSSRV. 
> 
> imho it's a bug when FS is not using the same nonce in the response as it received in the request. It might be a design flaw only that only appears when running against multiple servers through DNSSRV, but it's still a bug.
> 
> I'm sure it's easy to fix for Anthony with his m4d sk1llz ;)
> 




From mike at jerris.com  Wed Mar 10 01:56:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 04:56:33 -0500
Subject: [Freeswitch-users] [ERR] switch_rtp.c:2196
	FailedDTMFpayloadcheck.
In-Reply-To: <282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca>
	<36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com>
	<282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
Message-ID: <82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>

curious what happens if you change the level or payload type int hat config below as well.

Mike

On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:

> Some more info . . .
>  
> The error only seems to occur when a "0" is entered.
>  
> Looking at the Phone config, the RFC2833 options are :-
>  
> In Band
> Out of Band (RFC2833) - currently selected
> SIP Info
>  
> RTP Payload Type (96-127) - 96 currently selected
> RTP DTMP Level (0-63) - 0 Currently selected
>  
> Dave
> ----- Original Message -----
> From: Dave Stevenson
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:23 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.
> 
> Hi Mathieu,
>  
> if you tell me how to generate one, I'll do it now ?
>  
> regards
> Dave
> ----- Original Message -----
> From: Mathieu Rene
> To: freeswitch-users at lists.freeswitch.org
> Sent: Wednesday, March 03, 2010 8:09 PM
> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.
> 
> I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:
> 
>> Hi,
>>  
>> thanks Brian.
>>  
>> The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,
>>  
>> regards
>> Dave
>> ----- Original Message -----
>> From: Brian West
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Wednesday, March 03, 2010 7:50 PM
>> Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.
>> 
>> It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.
>> 
>> /b
>> 
>> On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:
>> 
>>>  
>>> It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?
>>>  
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Mar 10 02:00:20 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 05:00:20 -0500
Subject: [Freeswitch-users] SUCCESS!!!
In-Reply-To: <43146.92621.qm@web44811.mail.sp1.yahoo.com>
References: <E75CCC20-4E27-46D9-8A94-E273C04FA9AF@freeswitch.org>
	<407604F34A7540C3B971AABC4D952060@MOBILEE1705>
	<43146.92621.qm@web44811.mail.sp1.yahoo.com>
Message-ID: <A13FDADE-EE1C-4B1E-A5E9-1EA547F2C201@jerris.com>

I notice that you properly indented this, so I guess this runs in python?


On Mar 4, 2010, at 4:10 PM, Jim Thomas wrote:

> > PYTHON SUCKS
>  
> That's like saying:
>  
>   Asterisk sucks.
>   Asterisk is written in C language.
>   Therefore, C language sucks.
>  
> :-)
> 
>  
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, March 04, 2010 9:52 AM
> Subject: [Freeswitch-users] SUCCESS!!!
> 
> 
> >I have to say PYTHON SUCKS... ie Mailman is evil.
> > 

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From mike at jerris.com  Wed Mar 10 02:01:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Mar 2010 05:01:46 -0500
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
	<7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
Message-ID: <F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>

What pri dialect is this?

On Mar 4, 2010, at 11:11 PM, lakshmanan ganapathy wrote:

> Ok. This is to say how the problem got solved.
> Need openzap 1047 or above version.
> Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz 
> My telco is not accepting the display IE. 
> Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.
> 
> Thanks for all.
> 
> On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn <tomek.augustyn at gmail.com> wrote:
> Hello,
> 
> I had similar problem and I think it is more a problem between Sangoma card and your E1 provider than with freeswitch.
> 
> In my case it was necessary to set "origination_caller_id_number" to one of the telephone numbers linked to my E1 line. In other case the calls were rejected with "invalid information element" error.
> 
> You can try Sangoma's support they are very helpful.
> 
> Tomasz Augustyn

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From dome at tel.co.th  Wed Mar 10 02:12:58 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 10 Mar 2010 17:12:58 +0700
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>
	<5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
Message-ID: <8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>

2010/3/10 Michael Jerris <mike at jerris.com>:
> -nf has nothing to do with the performance, it has to do if the command
> blocks or not when you run it. ?I am unsure what you mean about sql in this
> context, it is totally unrelated. What is a cash db?
sorry i mean switch_cache_db_connection :)
i have problem about db connection. now switch_cache_db_connection use
1 connection for 1 thread
if i use mod_limit with odbc my server can't handle more than 100 connection.
So i'll try FS in no fork. but i want to know other effect.

BG

Dome C.
> Mike
>
> On Mar 10, 2010, at 12:10 AM, Dome Charoenyost wrote:
>
> Dear All,
> ??????????????How about performance when start FS with -nf ?(no
> fork). I ask because i want to keep SQL connect (cash db) in ?1
> connection. but i want to know other effect like a performance ,
> memory usage .. etc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jbrucehopkins at gmail.com  Wed Mar 10 04:00:31 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 12:00:31 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <000001cabfbb$f2114c90$d633e5b0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
Message-ID: <d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>

Hi Roly,

I hope to be able to have a go at this later today.  I'll let you know what
happens.

I have my FS box behind a hardware router/firewall, so I'm assuming I just
forward SIP port 5060 from the WAN to the FS box and use internal profile,
instead of using external profile with port 5080 forwarded as I have at the
moment.  We'll see what happens.

cheers
Bruce

On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at gilligan.id.au  Wed Mar 10 04:02:39 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Wed, 10 Mar 2010 23:02:39 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com> 
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
Message-ID: <9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>

Thanks for the idea i will try that next. I thought i would try to capture
the digits like normal first to make sure everything is working right and to
test a few event capture methods.  What i found was that mod_managed never
seemed to capture or read any digits for an out going call.  If i pass the
call to a different application/dial plan it captures the digits perfectly.
 I assume audio was working as it played audio to the user to enter digits
so i figure i have not configured something correctly but i have not idea
what it is.  Until i work it out i don't think any event code will work.

I have no idea what is wrong in this case or even how to debug it.

Chris

On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> One way might be to create an event loop and listen for DTMF
>
> public bool Load()
>         {
>             ThreadPool.QueueUserWorkItem((o) =>
>             {
>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>                 EventConsumer con = new EventConsumer("DTMF", "");
>                 while (true)
>                 {
>                     Event ev = con.pop(1);
>                     if (ev != null)
>                     {
>                                   ... do what ever needs to be done
>                     }
>
>                 }
>             });
>
> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>
> I am guessing the DTMF is "DTMF" so play around with that if it does not
> work.
>
> There are probably better ways.
>
>
>
> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> What i need to do is listen on a session before, during and after the
>> session is bridged with another call. I know built into FS is the option to
>> listen for *x where x is 0-9 but i really need more than one digit.
>>
>> I traced the method that listens for the single digit to the
>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>> switch_ivr_async.c
>>
>> mod_managed also has the following in the native class freeswitch.
>> public static switch_status_t
>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>> recv_dtmf)
>>
>> The problem is i can't work out how to use it.  I am fine the the first
>> param for the session but have no idea what i need for
>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>> recv_dtmf param.
>>
>> if anyone can point me in the right direction or knows a better way i
>> would be grateful.
>>
>> Chris
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Wed Mar 10 04:07:32 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 13:07:32 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
Message-ID: <65d96fc81003100407o4c55b3f3y1ba35c6fcfe5dc66@mail.gmail.com>

On Mon, Mar 8, 2010 at 11:02 PM, Brian West <brian at freeswitch.org> wrote:

> Thats unacceptable!!!  Wait its faxing... can't we just let it DIE
> already??  They have this thing called email... you should check it out...
> works great... even lets you attach video, audio and color images.... oh and
> text... dreary old text.... but hey you can mark it all up in HTML and fancy
> fonts now and make it anoying^H^H^H^H^H^H^Hpretty.
>
> /b
>
>
well Brian, thats a wrong assumption.... fax (T.30) is, and it will be,
present all over the world... you cannot do anything against (not even thing
about eradicate) it at least for the next 5 years :)

so, you will ave to comply ... we will have to comply if our customers keep
sending faxes :)


T.
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From tculjaga at gmail.com  Wed Mar 10 04:10:57 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 10 Mar 2010 13:10:57 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
Message-ID: <65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>

they just want to believe it is :) .. thats it.


T.

On Mon, Mar 8, 2010 at 11:46 PM, Brian West <brian at freeswitch.org> wrote:

> There is no such thing as security when humans are involved in the relay of
> information.  Thats fact.
>
> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
> involved its not secure.
>
> /b
>
> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>
> > While I'd love to see FAXing just die, especially with the amount of
> headache associated with it, the reality is that specific organizations
> still rely on it heavily on a daily basis (ie Health Care and
> Banking/Finance Industry). Plus they can legally get away with FAXing
> personal data directly between machines, since the FAX transmission doesn't
> seem to be directly covered by all their respective privacy regultions (ie
> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
> of these organizations just don't have the internal expertise to setup
> end-to-end encrypted email nor the impetus to deal with it, since FAXing
> works fine in their eyes. I think it's going to be around quite a while
> longer unless they actually stop making the FAX machines and force people to
> use something else.
> >
> > Cheers,
> > Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Wed Mar 10 05:42:34 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 10 Mar 2010 08:42:34 -0500
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com>
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com>
	<9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com>
Message-ID: <367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>

There should be no problem on digit capture on an outgoing call. For example
- when I originate a call, I ask the called party, for example, to press 1
to accept the call. This works correctly:

BLegSession.DtmfReceivedFunction = (d, t) =>
                {
                    Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d,
t);
                    if (d == confirmationDigit)
                    {
                        confirmed = true;
                        return "break";
                    }

                    return "";
                };

                while (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady() && loop <= loops)
                {
                    loop++;

                    if (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady())

BLegSession.StreamFile("prompts/press-1-to-accept-call-from.wav", 0);

                    if (!confirmed && BLegSession != null &&
BLegSession.Ready() && BLegSession.mediaReady())
                        BLegSession.CollectDigits(5000);

                }

If you are not receiving the digits - I would try and bridge the oubound
call to an existing call and check that you are getting two way audio - and
if you are using inband DTMF -  that you can actually hear the digits.

HTH


On Wed, Mar 10, 2010 at 7:02 AM, Chris <freeswitch at gilligan.id.au> wrote:

> Thanks for the idea i will try that next. I thought i would try to capture
> the digits like normal first to make sure everything is working right and to
> test a few event capture methods.  What i found was that mod_managed never
> seemed to capture or read any digits for an out going call.  If i pass the
> call to a different application/dial plan it captures the digits perfectly.
>  I assume audio was working as it played audio to the user to enter digits
> so i figure i have not configured something correctly but i have not idea
> what it is.  Until i work it out i don't think any event code will work.
>
> I have no idea what is wrong in this case or even how to debug it.
>
> Chris
>
>
> On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> One way might be to create an event loop and listen for DTMF
>>
>> public bool Load()
>>         {
>>             ThreadPool.QueueUserWorkItem((o) =>
>>             {
>>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>>                 EventConsumer con = new EventConsumer("DTMF", "");
>>                 while (true)
>>                 {
>>                     Event ev = con.pop(1);
>>                     if (ev != null)
>>                     {
>>                                   ... do what ever needs to be done
>>                     }
>>
>>                 }
>>             });
>>
>> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>>
>> I am guessing the DTMF is "DTMF" so play around with that if it does not
>> work.
>>
>> There are probably better ways.
>>
>>
>>
>> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>
>>> What i need to do is listen on a session before, during and after the
>>> session is bridged with another call. I know built into FS is the option to
>>> listen for *x where x is 0-9 but i really need more than one digit.
>>>
>>> I traced the method that listens for the single digit to the
>>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>>> switch_ivr_async.c
>>>
>>> mod_managed also has the following in the native class freeswitch.
>>> public static switch_status_t
>>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>> recv_dtmf)
>>>
>>> The problem is i can't work out how to use it.  I am fine the the first
>>> param for the session but have no idea what i need for
>>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>> recv_dtmf param.
>>>
>>> if anyone can point me in the right direction or knows a better way i
>>> would be grateful.
>>>
>>> Chris
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ahmed.ajmal at breezecom.ae  Wed Mar 10 06:01:38 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Wed, 10 Mar 2010 19:01:38 +0500
Subject: [Freeswitch-users] Set/Export channel variable
Message-ID: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>

Hi 

 

I am trying to set a channel variable on the bridge application using this:

 

<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>

 

The application runs fine, what I now need to do is set/export the 'gw'
channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.

 

Thanks

Ahmed

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From brian at freeswitch.org  Wed Mar 10 06:08:25 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Mar 2010 08:08:25 -0600
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
References: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
Message-ID: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>

We already set you a variable called sip_gateway_name when a call is placed out a gateway.  You're also doing this wrong you should NEVER EVER do @$ipgwX as that data is in the gateway.  You should simply do sofia/gateway/gwname/number

/b

On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:

> Hi
>  
> I am trying to set a channel variable on the bridge application using this:
>  
> <action application="bridge" data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 | [gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>
>  
> The application runs fine, what I now need to do is set/export the ?gw? channel variable for CDR so that I know which gateway was dialed. I have tried using set and export application but that doesn?t work. Please help.
>  
> Thanks
> Ahmed

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From srinivas.ksvreddy at gmail.com  Wed Mar 10 06:39:45 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Wed, 10 Mar 2010 09:39:45 -0500
Subject: [Freeswitch-users] forwarding incoming call from gateway to local
	externsion
Message-ID: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>

Hi Goodmorning.

i have connected my freeswtich to cisco gateway, i have registerd a
user(1099) in my freeswitch, and i have configured freeswitch any call from
1099 routing to cisco gateway, i need same way when call coming from gateway
with 101010 i want to route or farward to 1099, what files need to configure
any idea

Regard
srinvias
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From anthony.minessale at gmail.com  Wed Mar 10 08:25:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 10:25:46 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
Message-ID: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>

FS now tries to determine how accurate the timing is by doing some tests
with 1ms sleeps.
Most 64 bit CentOS box I have seen can easily calibrate.
If you are having that error alot but the calls still sound ok, it's safe to
ignore it.
you can also start FS with -nocal to skip this test.

Also make sure you do not have any other cpu-intensive application on your
box possibly competing for resources.


On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>
> what is this related to ?
>
> T.
>
>
> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>
>> Hi Anthony,
>>
>> Many, many apologies to take so long to reply - I failed to notice your
>> response.  Sorry !
>>
>> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
>> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
>> then does not give a warning.
>>
>> Best wishes
>> Bruce
>>
>>
>> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>>
>>> we are still wary about 5.3 due to bugs reported in libc.
>>> What number does it say it detected for the gap?
>>>
>>>
>>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com
>>> > wrote:
>>>
>>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>>> ....
>>>>
>>>>
>>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I wonder if anyone would be able to advise please:
>>>>>
>>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when
>>>>> I start FreeSWITCH that
>>>>>
>>>>> "Abnormally large timer gap detected"
>>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>>> experience audio problems".
>>>>>
>>>>> I get no such warning if I build on CentOS 5.3, and the test timings it
>>>>> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
>>>>> to Centos5.4 was a yum update on the 5.3 build.
>>>>>
>>>>> I guess the warning comes from here:
>>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>>
>>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>>> optiplex p4's I use for testing.
>>>>>
>>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or use
>>>>> 5.4 and not worry about the warnings, or if there is something I can do to
>>>>> fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>>> use such crummy hardware?!
>>>>>
>>>>> Many thanks in advance
>>>>> Bruce
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 10 09:09:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 11:09:46 -0600
Subject: [Freeswitch-users] Bug in mod_commands.c with show calls command
In-Reply-To: <alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003011607210.3234@osgood.microoffice.com>
	<191c3a031003011524o618a139frccab554c1c97d334@mail.gmail.com>
	<alpine.LRH.2.00.1003012226040.3234@osgood.microoffice.com>
Message-ID: <191c3a031003100909t37e97e92r2729dc6470d7e6d6@mail.gmail.com>

you can change the delimiter

Like the example:

"show calls as csv :::"

The data will be sep by ::: instead of comma

if you would like to make a patch to add "" it would be applied.



On Mon, Mar 1, 2010 at 9:29 PM, Wellie Chao <wchao at yahoo.com> wrote:

> Terrific. "show calls as xml" works great and just what I needed.
>
> P.S. "show calls as csv" still has the comma bug. Not a big deal since I
> can get it as XML. CSV needs quotes around field values that contain a
> comma. Pretty easy to fix. Few lines of C (strchr for "," and if found,
> put quotes around field value).
>
>
> Date: Mon, 1 Mar 2010 17:24:16 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Bug in mod_commands.c with show calls
> command
>
> Perhaps you may be reading up on something quickly and not worrying about
> the absolute correctness of something that is not critical to our time to
> explain.
>
> The proper syntax for the command is "show calls as xml" and you are not
> really using XMLRPC you are hitting it via direct url.
> so you will then need.
>
> http://192.168.1.1:8080/txtapi/show?calls%20as%20xml
>
>
> anything in the show command can be replied to with "as xml"
>
> also, for fun, try
>
> show calls as csv :::
>
>
>
>
> On Mon, Mar 1, 2010 at 3:31 PM, Wellie Chao <wchao at yahoo.com> wrote:
>      I am using the xml_rpc interface like so:
>
>      http://192.168.1.1:8080/webapi/show?calls
>
>      I get a response in a table, which is a little bit undesirable because
> I
>      want to parse the result in a program and don't want to parse a bunch
> of
>      HTML (just seems inelegant and wasteful), so I instead tried the
> following
>      request:
>
>      http://192.168.1.1:8080/txtapi/show?calls
>
>      That works better, but then another wrinkle appears. Caller ID (and
> Callee
>      ID) names sometimes have commas, which messes up the CSV.
>
>      This appears to be a bug in mod_commands.c (or I suppose you could
> call it
>      an artifact of somebody coding up something quickly and not worrying
> about
>      the absolute correctness of something that is not critical to the
> core).
>      There is also another bug with show calls via webapi in that the
> generated
>      HTML has spurious <td> tags. I think the spurious <td> tag bug arises
> due
>      to lines 3145-6 in mod/applications/mod_commands/mod_commands.c.
> Finally,
>      another bug (or maybe just unimplemented feature) is that xmlapl isn't
>      really different from webapi.
>
>      These problems occur in the latest source tree (20100301).
>
>      I am wondering if (a) one of the developers can fix these or (b) if I
> fix
>      the bugs, how can I submit patches [and if I submit patches, will they
> be
>      accepted into the main tree]?
>
>      The fixes are pretty trivial and I'd be happy to code them up if
> somebody
>      will tell me how I can submit the patches (haven't done it before).
>
>      _______________________________________________
>      FreeSWITCH-users mailing list
>      FreeSWITCH-users at lists.freeswitch.org
>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>      http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 10 09:11:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 11:11:41 -0600
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
Message-ID: <191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>

We have a mod_fax for t30 and work on t38 is underway.
So we already acknowledge that I guess.




On Wed, Mar 10, 2010 at 6:10 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> they just want to believe it is :) .. thats it.
>
>
> T.
>
>
> On Mon, Mar 8, 2010 at 11:46 PM, Brian West <brian at freeswitch.org> wrote:
>
>> There is no such thing as security when humans are involved in the relay
>> of information.  Thats fact.
>>
>> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
>> involved its not secure.
>>
>> /b
>>
>> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>>
>> > While I'd love to see FAXing just die, especially with the amount of
>> headache associated with it, the reality is that specific organizations
>> still rely on it heavily on a daily basis (ie Health Care and
>> Banking/Finance Industry). Plus they can legally get away with FAXing
>> personal data directly between machines, since the FAX transmission doesn't
>> seem to be directly covered by all their respective privacy regultions (ie
>> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
>> of these organizations just don't have the internal expertise to setup
>> end-to-end encrypted email nor the impetus to deal with it, since FAXing
>> works fine in their eyes. I think it's going to be around quite a while
>> longer unless they actually stop making the FAX machines and force people to
>> use something else.
>> >
>> > Cheers,
>> > Gabe
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From chris at fowler.cc  Wed Mar 10 09:16:10 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 10 Mar 2010 12:16:10 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only
	one	device ring
In-Reply-To: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>

Hi Lloyd,

You need to use sofia_contact in your bridge command in the dialplan:

e.g.
<anti-action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

This will cause FS to try all registrations simultaneously.

Cheers, Chris.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius Lloyd
Sent: Tuesday, March 09, 2010 11:53 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd
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From stevendt at primrosebank.net  Wed Mar 10 09:44:03 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 10 Mar 2010 17:44:03 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>
Message-ID: <C1A1AC0759924ED19A5FD2ABF82A9D2E@bp1.ad.bp.com>


  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 10, 2010 9:56 AM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  curious what happens if you change the level or payload type int hat config below as well.


  Mike


  On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


        Mathieu Rene
        Avant-Garde Solutions Inc
        Office: + 1 (514) 664-1044 x100
        Cell: +1 (514) 664-1044 x200
        mrene at avgs.ca








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From stevendt at primrosebank.net  Wed Mar 10 09:50:24 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 10 Mar 2010 17:50:24 -0000
Subject: [Freeswitch-users] [ERR]
	switch_rtp.c:2196FailedDTMFpayloadcheck.
References: <AD6C692525A0450C8AE2F3005B6D703B@bp1.ad.bp.com><59198F0C-C8CF-41AA-9489-62C297B5F90D@freeswitch.org><E6042A93C8D549E9A91335636CEBFAB4@bp1.ad.bp.com><3778B4E1-7DF7-4A11-BC1C-4476319EDAC3@avgs.ca><36BDEB9F2E174CA99548A71E14B21AA7@bp1.ad.bp.com><282476B55FD4450A8DBBAFCB673DCBEC@bp1.ad.bp.com>
	<82256D47-83CC-4BB3-98FD-48943769F621@jerris.com>
Message-ID: <B8B7700F9FBA4CE3B1B6F059307C88CE@bp1.ad.bp.com>

Mike,

thanks for following this up . . . . .

Not having the faintest idea what those settings might do, I never thought of changing them and see what difference it made, but you prompted me to try that....


I've still no idea what "RTP Payload Type" is, but changing it makes no difference to my problem.

However, DTMF level (the "DTMP" was a typo in my earlier message), which I'm guessing is a kind of volume setting (?) DOES make a difference.

Setting it to different values makes the error go away, it's currently set at "1" (from "0") which seems to have been enough to get rid of the problem. Strange that the problem only affected one digit ("0"), but maybe something was just on a limit somewhere - maybe the phone has an issue with certain frequencies.

Anyway, a simple work around, if only I'd had the sense to try that earlier - thanks a lot for the prompt !

regards
Dave


  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 10, 2010 9:56 AM
  Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196FailedDTMFpayloadcheck.


  curious what happens if you change the level or payload type int hat config below as well.


  Mike


  On Mar 3, 2010, at 3:39 PM, Dave Stevenson wrote:


    Some more info . . .

    The error only seems to occur when a "0" is entered.

    Looking at the Phone config, the RFC2833 options are :-

    In Band
    Out of Band (RFC2833) - currently selected
    SIP Info

    RTP Payload Type (96-127) - 96 currently selected
    RTP DTMP Level (0-63) - 0 Currently selected

    Dave
      ----- Original Message -----
      From: Dave Stevenson
      To: freeswitch-users at lists.freeswitch.org
      Sent: Wednesday, March 03, 2010 8:23 PM
      Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 FailedDTMFpayloadcheck.


      Hi Mathieu,

      if you tell me how to generate one, I'll do it now ?

      regards
      Dave
        ----- Original Message -----
        From: Mathieu Rene
        To: freeswitch-users at lists.freeswitch.org
        Sent: Wednesday, March 03, 2010 8:09 PM
        Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMFpayloadcheck.


        I don't recall any rfc2833 changes in the past 2 weeks. I would like to see a packet capture of those bogus rtp packets though.


        Mathieu Rene
        Avant-Garde Solutions Inc
        Office: + 1 (514) 664-1044 x100
        Cell: +1 (514) 664-1044 x200
        mrene at avgs.ca








        On 3-Mar-10, at 3:06 PM, Dave Stevenson wrote:


          Hi,

          thanks Brian.

          The device is a Thomson ST2030 which has a basic listing on the Interop Wiki page, but does not specifically mention rfc2833 (although rtp is marked as "not tested"). I have not seen this message before while running the FS version that I'm currently using 16543 - yes, I know that this is a few weeks old and I will update and/or raise a bug once I've upgraded, but it would be nice to know if anyone else sees problems with this phone,

          regards
          Dave
            ----- Original Message -----
            From: Brian West
            To: freeswitch-users at lists.freeswitch.org
            Sent: Wednesday, March 03, 2010 7:50 PM
            Subject: Re: [Freeswitch-users] [ERR] switch_rtp.c:2196 Failed DTMF payloadcheck.


            It usually means your device can't do rfc2833 correctly and needs to have a bug opened so they can fix it.


            /b


            On Mar 3, 2010, at 1:39 PM, Dave Stevenson wrote:



              It appears in response to keypad events during voicemail retrieval, the prompts were actioned correctly, but I was wondering what the message actually means ?






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From peder at networkoblivion.com  Wed Mar 10 11:27:03 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 10 Mar 2010 13:27:03 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <019801cac087$a9da3b90$fd8eb2b0$@com>

Is there a way to tell how accurate the timing is on a box before running
FreeSWITCH on it?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, March 10, 2010 10:26 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4

 

FS now tries to determine how accurate the timing is by doing some tests
with 1ms sleeps.
Most 64 bit CentOS box I have seen can easily calibrate.
If you are having that error alot but the calls still sound ok, it's safe to
ignore it.
you can also start FS with -nocal to skip this test.

Also make sure you do not have any other cpu-intensive application on your
box possibly competing for resources.



On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...

what is this related to ?

T.

 

On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
wrote:

Hi Anthony,

Many, many apologies to take so long to reply - I failed to notice your
response.  Sorry !

It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
then does not give a warning.

Best wishes
Bruce

 

On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

we are still wary about 5.3 due to bugs reported in libc.
What number does it say it detected for the gap?



On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
wrote:

OK - I've realised I do get the same warning with CentOS 5.3, it just goes
past more quickly so I didn't see it.  Maybe it is just the hardware ....

 

On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

Hi,

I wonder if anyone would be able to advise please:

When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
start FreeSWITCH that 

"Abnormally large timer gap detected"
"Do you have your kernel timer set to greater than 1kHz?  You may experience
audio problems".

I get no such warning if I build on CentOS 5.3, and the test timings it
measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
to Centos5.4 was a yum update on the 5.3 build.

I guess the warning comes from here:
http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time
.c
<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_t
ime.c> 

This is all on pretty low spec hardware - a couple of different Dell
optiplex p4's I use for testing.

Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
and not worry about the warnings, or if there is something I can do to fix
the problem it is warning about.  Perhaps it is just that I shouldn't use
such crummy hardware?!

Many thanks in advance
Bruce

 

 

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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900


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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
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From jbrucehopkins at gmail.com  Wed Mar 10 11:38:21 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 19:38:21 +0000
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <d03a10e01003101138n65918b8bx6798d0860a982d12@mail.gmail.com>

Many thanks for the info.  I was using a fairly crummy 32 bit box with no
PSTN card installed.  There was no degredation of the audio immediately
apparent.

Bests
Bruce

On 10 March 2010 16:25, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>
>
> On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>>
>> what is this related to ?
>>
>> T.
>>
>>
>> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>
>>> Hi Anthony,
>>>
>>> Many, many apologies to take so long to reply - I failed to notice your
>>> response.  Sorry !
>>>
>>> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
>>> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
>>> then does not give a warning.
>>>
>>> Best wishes
>>> Bruce
>>>
>>>
>>> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>wrote:
>>>
>>>> we are still wary about 5.3 due to bugs reported in libc.
>>>> What number does it say it detected for the gap?
>>>>
>>>>
>>>> On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <
>>>> jbrucehopkins at gmail.com> wrote:
>>>>
>>>>> OK - I've realised I do get the same warning with CentOS 5.3, it just
>>>>> goes past more quickly so I didn't see it.  Maybe it is just the hardware
>>>>> ....
>>>>>
>>>>>
>>>>> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com>wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I wonder if anyone would be able to advise please:
>>>>>>
>>>>>> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning
>>>>>> when I start FreeSWITCH that
>>>>>>
>>>>>> "Abnormally large timer gap detected"
>>>>>> "Do you have your kernel timer set to greater than 1kHz?  You may
>>>>>> experience audio problems".
>>>>>>
>>>>>> I get no such warning if I build on CentOS 5.3, and the test timings
>>>>>> it measures on starting FreeSWITCH do look lower.  All I was doing to
>>>>>> upgrade to Centos5.4 was a yum update on the 5.3 build.
>>>>>>
>>>>>> I guess the warning comes from here:
>>>>>> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>>>>>>
>>>>>> This is all on pretty low spec hardware - a couple of different Dell
>>>>>> optiplex p4's I use for testing.
>>>>>>
>>>>>> Does anyone happen to know if I should just stick to Cent)S 5.3, or
>>>>>> use 5.4 and not worry about the warnings, or if there is something I can do
>>>>>> to fix the problem it is warning about.  Perhaps it is just that I shouldn't
>>>>>> use such crummy hardware?!
>>>>>>
>>>>>> Many thanks in advance
>>>>>> Bruce
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 10 11:44:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Mar 2010 11:44:43 -0800
Subject: [Freeswitch-users] forwarding incoming call from gateway to
	local externsion
In-Reply-To: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>
References: <f8af5741003100639k14d24e2brfda2ff4bce61e589@mail.gmail.com>
Message-ID: <87f2f3b91003101144w31dc79b3x27c1d5e5d6f8b449@mail.gmail.com>

On Wed, Mar 10, 2010 at 6:39 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi Goodmorning.
>
> i have connected my freeswtich to cisco gateway, i have registerd a
> user(1099) in my freeswitch, and i have configured freeswitch any call from
> 1099 routing to cisco gateway, i need same way when call coming from gateway
> with 101010 i want to route or farward to 1099, what files need to configure
> any idea
>

Look in conf/dialplan/public.xml and see how the default extension numbers
(1000-1019) are handled in this scenario.
-MC
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From anthony.minessale at gmail.com  Wed Mar 10 12:29:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 14:29:02 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <019801cac087$a9da3b90$fd8eb2b0$@com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
	<019801cac087$a9da3b90$fd8eb2b0$@com>
Message-ID: <191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>

The best tool we have is actually running freeswitch on it to see.


On Wed, Mar 10, 2010 at 1:27 PM, Peder <peder at networkoblivion.com> wrote:

>  Is there a way to tell how accurate the timing is on a box before running
> FreeSWITCH on it?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Wednesday, March 10, 2010 10:26 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4
>
>
>
> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>  On Wed, Mar 10, 2010 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
> i have the same issue but on centos 4.8 ... on 5.4 doesn't complain...
>
> what is this related to ?
>
> T.
>
>
>
> On Mon, Mar 8, 2010 at 2:25 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
> wrote:
>
> Hi Anthony,
>
> Many, many apologies to take so long to reply - I failed to notice your
> response.  Sorry !
>
> It seems to be variable.  Most times (on CentOS 5.3) when I start FS it
> tells me it is around 1995ms.  Occasionally it is lower (sub 1000ms) and
> then does not give a warning.
>
> Best wishes
> Bruce
>
>
>
> On 1 March 2010 16:47, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> we are still wary about 5.3 due to bugs reported in libc.
> What number does it say it detected for the gap?
>
>   On Sun, Feb 28, 2010 at 12:03 PM, Bruce Hopkins <jbrucehopkins at gmail.com>
> wrote:
>
>  OK - I've realised I do get the same warning with CentOS 5.3, it just
> goes past more quickly so I didn't see it.  Maybe it is just the hardware
> ....
>
>
>
> On 28 February 2010 15:37, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
> Hi,
>
> I wonder if anyone would be able to advise please:
>
> When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I
> start FreeSWITCH that
>
> "Abnormally large timer gap detected"
> "Do you have your kernel timer set to greater than 1kHz?  You may
> experience audio problems".
>
> I get no such warning if I build on CentOS 5.3, and the test timings it
> measures on starting FreeSWITCH do look lower.  All I was doing to upgrade
> to Centos5.4 was a yum update on the 5.3 build.
>
> I guess the warning comes from here:
> http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_time.c<http://fisheye.freeswitch.org/browse/%7Eraw,r=16409/FreeSWITCH/src/switch_time.c>
>
> This is all on pretty low spec hardware - a couple of different Dell
> optiplex p4's I use for testing.
>
> Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4
> and not worry about the warnings, or if there is something I can do to fix
> the problem it is warning about.  Perhaps it is just that I shouldn't use
> such crummy hardware?!
>
> Many thanks in advance
> Bruce
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From siniypin at gmail.com  Wed Mar 10 12:52:00 2010
From: siniypin at gmail.com (RobertT)
Date: Wed, 10 Mar 2010 23:52:00 +0300
Subject: [Freeswitch-users] sofia_contact strange behavior
Message-ID: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>

Hello everybody!

I am experiencing problems with sofia_contact command.
When I dial sofia_contact external/1000@<domain> in fs cli I see correct
registration data, provided the user 1000 is registered.
But in the same time when I call
${sofia_conatct(external/${dialed_ext}@$${domain})}
from dialplan as an argument to deflect application I recieve
error\user_not_registered.
What the hell?

Best regards, RobertT.
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From peder at networkoblivion.com  Wed Mar 10 12:53:47 2010
From: peder at networkoblivion.com (Peder)
Date: Wed, 10 Mar 2010 14:53:47 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
Message-ID: <021b01cac093$c76d3930$5647ab90$@com>

OK, can you decode what this means exactly?  I'd be happy to write something
up to reference users to, but I don't really know what it means.  It is a a
quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
150, but is that good, bad, indifferent?  I've tried to google it, but
without really knowing what it is I am looking for, I am not getting good
results.

 

 

2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating timer,
please wait...

2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000 Average:
2001 Step: 50

2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
1998 Step: 50

2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
1998 Step: 50

2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
1998 Step: 50

2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
1998 Step: 50

2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
1998 Step: 50

2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
1998 Step: 50

2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
1997 Step: 50

2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
1998 Step: 50

2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
1998 Step: 50

2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
1998 Step: 50

2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
1998 Step: 50

2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
1998 Step: 50

2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
1018 Step: 50

2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
998 Step: 50

2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
1018 Step: 50

2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
998 Step: 50

2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
1018 Step: 50

2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
998 Step: 50

2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
1038 Step: 50

2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
998 Step: 50

2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
1018 Step: 50

2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
998 Step: 50

2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
1018 Step: 50

2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
Step: 50

2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
1018 Step: 50

2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
Step: 50

2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average: 1018
Step: 50

2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000 Average:
2018 Step: 50

2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
1998 Step: 50

2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
1578 Step: 50

2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
998 Step: 50

2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of 150
calculated

2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [CORE_SOFTTIMER_MODULE]

2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
Timer 'soft'

2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [CORE_PCM_MODULE]

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From ben at langfeld.co.uk  Wed Mar 10 13:05:32 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 10 Mar 2010 21:05:32 +0000
Subject: [Freeswitch-users] Tone Detection During Bridge
Message-ID: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>

Dear List,

I have a requirement (due to poor quality Sipura products) to have
freeswitch listen for a tone during a bridge (standard local extension
dialplan segment). I have the required line:

<action appication="tone_detect" data="hangup 400 r 0 hangup
'normal_clearing' 1"/>

But I need to have this run _during_ the bridge (to listen for the tone
coming from the broken SPA3k and to break the bridge). Does anyone have any
idea how this may be achieved? There is no such documentation on the wiki.
This works perfectly if the bridge was not successful, and so will detect
tones during voicemail messaging.

Regards,
Ben Langfeld

Wave > Email
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From ledoktre at meanie.us  Wed Mar 10 13:26:15 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 10 Mar 2010 15:26:15 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
	<4B8569CB.6070804@meanie.us>
	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
Message-ID: <4B980E77.8080303@meanie.us>

Giovanni,

Any progress updates on the skypiax updates for sound quality and also 
DMTF detection?

With this update coming, I am pretty excited to be trying it out.  I 
have been trying very, very hard to be patient... :-D

Thanks,

Giovanni Maruzzelli wrote:
> On Wed, Feb 24, 2010 at 7:02 PM, Tim Streit <ledoktre at meanie.us> wrote:
>   
>> Hello,
>>
>> I was writing to inquire how this skypiax update was coming along.  I
>> didn't see it in the mailing list, but also since it had been nearly 1
>> week, I wanted to be sure if I didn't miss the announcement.  I am very
>> anxious to try this new update of the module.. It should be awesome!
>>     
>
> Hehehe, no, you've not missed the announcement.
>
> Is taking me some more time than I was expecting.
> But's arriving...
> I'll post here the announcement ;)
>
> -giovanni
>
>
>
>   
>> Thanks,
>>
>> Tim
>>
>> Giovanni Maruzzelli wrote:
>>     
>>> before to delve in the troubleshooting, I have to say that I'm
>>> modifying the audio skypiax code in svn, so maybe it's just my fault
>>> ;).
>>>
>>> please be patient for a little while, I hope to have done with it in a
>>> couple days.
>>>
>>> I'll announce to the mailing list when done.
>>>
>>> In the mean time, at least one good news for you user of SkypeIn
>>> service: a new feature of mod_skypiax is intended to recognize the
>>> DTMFs coming from SkypeIn, so the incoming calls will be able to use
>>> ivr, voicemail, etc.
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
>
>   



From jbrucehopkins at gmail.com  Wed Mar 10 13:28:30 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 21:28:30 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
Message-ID: <d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>

Hey Roly,

Are you getting anywhere?  I'm still struggling, I'm afraid.

I'm able to call from WAN --> LAN using the internal profile on port 5060,
but I still can't call out from LAN --> WAN so there is obviously something
really basic I'm still not getting.

In sip_profiles/internal.xml I have:

 <param name="aggressive-nat-detection" value="true"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>

<!-- the external_rtp_ip etc being defined in vars.xml -->


In this example I have left <param name="force-register-domain"
value="$${domain}"/> etc uncommented to force the registration to go to
default.xml in the dialplan.

I thought maybe this was my problem and tried putting a separate file in the
directory to enable the WAN phone to register using the external DNS name
for the server as its domain, and uncommenting force-register-domain, etc.
But then I got an error saying that the local (to Freeswitch) phone in the
LAN was not registered when I tried to make the call from the WAN.

Any thoughts?

cheers
Bruce






On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi Roly,
>
> I hope to be able to have a go at this later today.  I'll let you know what
> happens.
>
> I have my FS box behind a hardware router/firewall, so I'm assuming I just
> forward SIP port 5060 from the WAN to the FS box and use internal profile,
> instead of using external profile with port 5080 forwarded as I have at the
> moment.  We'll see what happens.
>
> cheers
> Bruce
>
> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>
>> Ah I see... what? Please share and lead this blind man out the FS
>> wilderness!
>>
>> I don't understand...what happens to the external profile? Do you delete
>> it?
>> And how do you forward port 5060?
>>
>> ...and you thought you were a newbie!
>>
>> Any insight would be much appreciated...loving the journey.
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
>> Hopkins
>> Sent: 09 March 2010 08:23 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>
>> Ah I see.  I will try again using the internal profile and forwarding
>> port 5060.  Presumably still creating a directory entry to enable the
>> outside-facing domain to be used.
>>
>> Many thanks for your patient help of a newbie Brian.
>>
>> Cheers
>> Bruce
>>
>> Please excuse the brevity - sent from my mobile.
>>
>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>
>> > ok you aren't catching one key thing here.. you no longer need two
>> > profiles.
>> >
>> > /b
>> >
>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>> >
>> >> Hi again,
>> >>
>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>> >> something wrong here as I am still having difficulty, and still
>> >> have the same problem.
>> >>
>> >> I updated to build 16938 by means of "make current"
>> >>
>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>> >> to use 5080 instead of 5090.
>> >>
>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>> >> the server as a domain for the remote softphone to register on.  I
>> >> left the group name in this entry the same as inthe default entry,
>> >> so that the remote phone could register on the same extension
>> >> numbers (100, etc) as in the default build.
>> >>
>> >> I still find that, if I initiate a call from the local (on same LAN
>> >> as freeswitch) phone to the remote phone, I get the message on the
>> >> CLI:
>> >>
>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>> >> type [user] cause: [USER_NOT_REGISTERED]
>> >>
>> >> One possibly unrelated aside, I also found I needed to uncomment
>> >> <param name="aggressive-nat-detection" value="true"/> in
>> >> external.xml, otherwise in the case of a call initiated by the
>> >> remote phone being hung up by the local phone, freeswitch sent the
>> >> BYE to the private IP of the remote phone, rather than its public
>> >> ip - meaning that the remote phone didn't receive the BYE.
>> >>
>> >> Any further ideas where I am going wrong here please?
>> >>
>> >> thanks again in advance
>> >> Bruce
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From freeswitch at gilligan.id.au  Wed Mar 10 13:42:40 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Thu, 11 Mar 2010 08:42:40 +1100
Subject: [Freeswitch-users] mod_managed - how listen for *999
In-Reply-To: <367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>
References: <9394d42f1003090227p1295f785o9c9c1a5d3f6e3805@mail.gmail.com> 
	<367751821003091831t36fe118btf811501a79b614d5@mail.gmail.com> 
	<9394d42f1003100402o574f606br2e59d45f9733a117@mail.gmail.com> 
	<367751821003100542j192503bgdbdacf9780269d6c@mail.gmail.com>
Message-ID: <9394d42f1003101342p55c26227xdeb298dec3f68d11@mail.gmail.com>

Thanks Phillip! That worked perfectly for some reason.  I mean it was hardly
different from mine and i still can't spot the fault in my own code.  very
strange.


On Thu, Mar 11, 2010 at 12:42 AM, Phillip Jones <pjintheusa at gmail.com>wrote:

> There should be no problem on digit capture on an outgoing call. For
> example - when I originate a call, I ask the called party, for example, to
> press 1 to accept the call. This works correctly:
>
> BLegSession.DtmfReceivedFunction = (d, t) =>
>                 {
>                     Log.WriteLine(LogLevel.Info, "Received {0} for {1}.",
> d, t);
>                     if (d == confirmationDigit)
>                     {
>                         confirmed = true;
>                         return "break";
>                     }
>
>                     return "";
>                 };
>
>                 while (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady() && loop <= loops)
>                 {
>                     loop++;
>
>                     if (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady())
>
> BLegSession.StreamFile("prompts/press-1-to-accept-call-from.wav", 0);
>
>                     if (!confirmed && BLegSession != null &&
> BLegSession.Ready() && BLegSession.mediaReady())
>                         BLegSession.CollectDigits(5000);
>
>                 }
>
> If you are not receiving the digits - I would try and bridge the oubound
> call to an existing call and check that you are getting two way audio - and
> if you are using inband DTMF -  that you can actually hear the digits.
>
> HTH
>
>
>
> On Wed, Mar 10, 2010 at 7:02 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Thanks for the idea i will try that next. I thought i would try to capture
>> the digits like normal first to make sure everything is working right and to
>> test a few event capture methods.  What i found was that mod_managed never
>> seemed to capture or read any digits for an out going call.  If i pass the
>> call to a different application/dial plan it captures the digits perfectly.
>>  I assume audio was working as it played audio to the user to enter digits
>> so i figure i have not configured something correctly but i have not idea
>> what it is.  Until i work it out i don't think any event code will work.
>>
>> I have no idea what is wrong in this case or even how to debug it.
>>
>> Chris
>>
>>
>> On Wed, Mar 10, 2010 at 1:31 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>> One way might be to create an event loop and listen for DTMF
>>>
>>> public bool Load()
>>>         {
>>>             ThreadPool.QueueUserWorkItem((o) =>
>>>             {
>>>                 Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
>>>                 EventConsumer con = new EventConsumer("DTMF", "");
>>>                 while (true)
>>>                 {
>>>                     Event ev = con.pop(1);
>>>                     if (ev != null)
>>>                     {
>>>                                   ... do what ever needs to be done
>>>                     }
>>>
>>>                 }
>>>             });
>>>
>>> See http://wiki.freeswitch.org/wiki/Event_List#DTMF
>>>
>>> I am guessing the DTMF is "DTMF" so play around with that if it does not
>>> work.
>>>
>>> There are probably better ways.
>>>
>>>
>>>
>>> On Tue, Mar 9, 2010 at 5:27 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>>
>>>> What i need to do is listen on a session before, during and after the
>>>> session is bridged with another call. I know built into FS is the option to
>>>> listen for *x where x is 0-9 but i really need more than one digit.
>>>>
>>>> I traced the method that listens for the single digit to the
>>>> command switch_core_event_hook_add_send_dtmf(session, meta_on_dtmf); in
>>>> switch_ivr_async.c
>>>>
>>>> mod_managed also has the following in the native class freeswitch.
>>>> public static switch_status_t
>>>> switch_core_event_hook_add_recv_dtmf(SWIGTYPE_p_switch_core_session session,
>>>> SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>>> recv_dtmf)
>>>>
>>>> The problem is i can't work out how to use it.  I am fine the the first
>>>> param for the session but have no idea what i need for
>>>> the SWIGTYPE_p_f_p_switch_core_session_p_q_const__switch_dtmf_t_enum_switch_dtmf_direction_t__switch_status_t
>>>> recv_dtmf param.
>>>>
>>>> if anyone can point me in the right direction or knows a better way i
>>>> would be grateful.
>>>>
>>>> Chris
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Wed Mar 10 13:52:29 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 16:52:29 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
Message-ID: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>

Chris, Thank you.

I find the following information from wiki.This is say sofia.conf. I think
sofia.conf.xml

Multiple Registrations

Call one extension and ring several phones

You must enable multiple registrations in sofia.conf

<param name="multiple-registrations" value="true"/>


----

Which file I should enable internal.xml or sofia.conf.xml

Thanks

Lloyd




On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Hi Lloyd,
>
>
>
> You need to use sofia_contact in your bridge command in the dialplan:
>
>
>
> e.g.
>
> *<anti-action application="bridge"
> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>
> * *
>
> This will cause FS to try all registrations simultaneously.
>
>
>
> Cheers, Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Tuesday, March 09, 2010 11:53 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
> device ring
>
>
>
> Hi All,
>
>
>
> I enable the multiple registrations in sip_profiles/internal.xml.
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> Then I restart the freeswitch.
>
>
>
> I use Aastra 9133i and eyebeam register for same extension.
>
>
>
> But randomly only one device ring. Do I need to enable any other settings?
>
>
>
>
> Thanks,
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jbrucehopkins at gmail.com  Wed Mar 10 14:04:58 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 22:04:58 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
	<d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
Message-ID: <d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>

Interestingly I *can* call outbound to the WAN a SIP client registered with
FS via Fring.  I wonder if Fring have some additional nat traversal/SBC
malarkey going on as they proxy the calls.

I can't call outbound to an Eyebeam softphone registered directly wth FS
though - although said Eyebeam is registered and can call inwards.

I must have my nat handling configured wrongly.

B

On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hey Roly,
>
> Are you getting anywhere?  I'm still struggling, I'm afraid.
>
> I'm able to call from WAN --> LAN using the internal profile on port 5060,
> but I still can't call out from LAN --> WAN so there is obviously something
> really basic I'm still not getting.
>
> In sip_profiles/internal.xml I have:
>
>
>  <param name="aggressive-nat-detection" value="true"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
> <!-- the external_rtp_ip etc being defined in vars.xml -->
>
>
> In this example I have left <param name="force-register-domain"
> value="$${domain}"/> etc uncommented to force the registration to go to
> default.xml in the dialplan.
>
> I thought maybe this was my problem and tried putting a separate file in
> the directory to enable the WAN phone to register using the external DNS
> name for the server as its domain, and uncommenting force-register-domain,
> etc.  But then I got an error saying that the local (to Freeswitch) phone in
> the LAN was not registered when I tried to make the call from the WAN.
>
> Any thoughts?
>
> cheers
> Bruce
>
>
>
>
>
>
>
> On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi Roly,
>>
>> I hope to be able to have a go at this later today.  I'll let you know
>> what happens.
>>
>> I have my FS box behind a hardware router/firewall, so I'm assuming I just
>> forward SIP port 5060 from the WAN to the FS box and use internal profile,
>> instead of using external profile with port 5080 forwarded as I have at the
>> moment.  We'll see what happens.
>>
>> cheers
>> Bruce
>>
>> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>>
>>> Ah I see... what? Please share and lead this blind man out the FS
>>> wilderness!
>>>
>>> I don't understand...what happens to the external profile? Do you delete
>>> it?
>>> And how do you forward port 5060?
>>>
>>> ...and you thought you were a newbie!
>>>
>>> Any insight would be much appreciated...loving the journey.
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Bruce
>>> Hopkins
>>> Sent: 09 March 2010 08:23 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>>
>>> Ah I see.  I will try again using the internal profile and forwarding
>>> port 5060.  Presumably still creating a directory entry to enable the
>>> outside-facing domain to be used.
>>>
>>> Many thanks for your patient help of a newbie Brian.
>>>
>>> Cheers
>>> Bruce
>>>
>>> Please excuse the brevity - sent from my mobile.
>>>
>>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>>
>>> > ok you aren't catching one key thing here.. you no longer need two
>>> > profiles.
>>> >
>>> > /b
>>> >
>>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>>> >
>>> >> Hi again,
>>> >>
>>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>>> >> something wrong here as I am still having difficulty, and still
>>> >> have the same problem.
>>> >>
>>> >> I updated to build 16938 by means of "make current"
>>> >>
>>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>>> >> to use 5080 instead of 5090.
>>> >>
>>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>>> >> the server as a domain for the remote softphone to register on.  I
>>> >> left the group name in this entry the same as inthe default entry,
>>> >> so that the remote phone could register on the same extension
>>> >> numbers (100, etc) as in the default build.
>>> >>
>>> >> I still find that, if I initiate a call from the local (on same LAN
>>> >> as freeswitch) phone to the remote phone, I get the message on the
>>> >> CLI:
>>> >>
>>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>>> >> type [user] cause: [USER_NOT_REGISTERED]
>>> >>
>>> >> One possibly unrelated aside, I also found I needed to uncomment
>>> >> <param name="aggressive-nat-detection" value="true"/> in
>>> >> external.xml, otherwise in the case of a call initiated by the
>>> >> remote phone being hung up by the local phone, freeswitch sent the
>>> >> BYE to the private IP of the remote phone, rather than its public
>>> >> ip - meaning that the remote phone didn't receive the BYE.
>>> >>
>>> >> Any further ideas where I am going wrong here please?
>>> >>
>>> >> thanks again in advance
>>> >> Bruce
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From gmaruzz at celliax.org  Wed Mar 10 14:20:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 10 Mar 2010 23:20:04 +0100
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <4B980E77.8080303@meanie.us>
References: <4B7F0302.3060303@meanie.us>
	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com> 
	<4B8569CB.6070804@meanie.us>
	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com> 
	<4B980E77.8080303@meanie.us>
Message-ID: <7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>

On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre at meanie.us> wrote:

> With this update coming, I am pretty excited to be trying it out. ?I
> have been trying very, very hard to be patient... :-D

Ciao Tim

hehehe, you've missed the announce, so!

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.html




-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From chris.chen2004 at gmail.com  Wed Mar 10 14:21:51 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Wed, 10 Mar 2010 17:21:51 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <507898381003101421j493a075cu7e07b14bd13133ae@mail.gmail.com>

Lloyd,
Just the one in internal.xml should do the work.


On Wed, Mar 10, 2010 at 4:52 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Chris, Thank you.
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
> Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
> <param name="multiple-registrations" value="true"/>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>
> Thanks
>
> Lloyd
>
>
>
>
> On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
>
>>  Hi Lloyd,
>>
>>
>>
>> You need to use sofia_contact in your bridge command in the dialplan:
>>
>>
>>
>> e.g.
>>
>> *<anti-action application="bridge"
>> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>>
>> * *
>>
>> This will cause FS to try all registrations simultaneously.
>>
>>
>>
>> Cheers, Chris.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
>> Lloyd
>> *Sent:* Tuesday, March 09, 2010 11:53 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
>> device ring
>>
>>
>>
>> Hi All,
>>
>>
>>
>> I enable the multiple registrations in sip_profiles/internal.xml.
>>
>>
>>
>> <param name="multiple-registrations" value="true"/>
>>
>>
>>
>> Then I restart the freeswitch.
>>
>>
>>
>> I use Aastra 9133i and eyebeam register for same extension.
>>
>>
>>
>> But randomly only one device ring. Do I need to enable any other settings?
>>
>>
>>
>>
>> Thanks,
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 10 14:23:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Mar 2010 14:23:01 -0800
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <87f2f3b91003101423n740fd293p227550ceaca82cfb@mail.gmail.com>

On Wed, Mar 10, 2010 at 1:52 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Chris, Thank you.
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
> Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
> <param name="multiple-registrations" value="true"/>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>

It appears commented out in internal.xml so that's a pretty good hint of
which file it goes in...
-MC
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From gmaruzz at celliax.org  Wed Mar 10 14:23:45 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 10 Mar 2010 23:23:45 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <021b01cac093$c76d3930$5647ab90$@com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com> 
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com> 
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com> 
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com> 
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com> 
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com> 
	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com> 
	<021b01cac093$c76d3930$5647ab90$@com>
Message-ID: <7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>

On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
> OK, can you decode what this means exactly?? I?d be happy to write something
> up to reference users to, but I don?t really know what it means.? It is a a
> quad-core 2.8GHz running CentOS 5.3 64-bit.? It looks like it calculated
> 150, but is that good, bad, indifferent?? I?ve tried to google it, but
> without really knowing what it is I am looking for, I am not getting good
> results.

This tells you the calculated average discrepancy of the timing is 150
microseconds (millisecond/1000), that I believe is good. (hope this
explanation is true ;) )


>
>
>
>
>
> 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating timer,
> please wait...
>
> 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2001 Step: 50
>
> 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
> 1997 Step: 50
>
> 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
> 1038 Step: 50
>
> 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 1018 Step: 50
>
> 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average: 998
> Step: 50
>
> 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average: 1018
> Step: 50
>
> 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2018 Step: 50
>
> 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
> 1998 Step: 50
>
> 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
> 1578 Step: 50
>
> 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> 998 Step: 50
>
> 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of 150
> calculated
>
> 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [CORE_SOFTTIMER_MODULE]
>
> 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
> Timer 'soft'
>
> 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
> Successfully Loaded [CORE_PCM_MODULE]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From chris at fowler.cc  Wed Mar 10 14:26:49 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 10 Mar 2010 17:26:49 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only
	one	device ring
In-Reply-To: <8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>

Hi Lloyd,

It depends on which profile your phones are registering with.  I suspect internal.xml will work.

Chris.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Aloysius Lloyd
Sent: Wednesday, March 10, 2010 1:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Chris, Thank you.

I find the following information from wiki.This is say sofia.conf. I think sofia.conf.xml

Multiple Registrations

Call one extension and ring several phones

You must enable multiple registrations in sofia.conf



<param name="multiple-registrations" value="true"/>



----

Which file I should enable internal.xml or sofia.conf.xml

Thanks

Lloyd



On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
Hi Lloyd,

You need to use sofia_contact in your bridge command in the dialplan:

e.g.
<anti-action application="bridge" data="${sofia_contact(${dialed_extension}@${domain_name})}"/>

This will cause FS to try all registrations simultaneously.

Cheers, Chris.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Aloysius Lloyd
Sent: Tuesday, March 09, 2010 11:53 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Multiple Registrations - randomly only one device ring

Hi All,

I enable the multiple registrations in sip_profiles/internal.xml.

<param name="multiple-registrations" value="true"/>

Then I restart the freeswitch.

I use Aastra 9133i and eyebeam register for same extension.

But randomly only one device ring. Do I need to enable any other settings?


Thanks,
Lloyd

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Mar 10 14:31:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 16:31:24 -0600
Subject: [Freeswitch-users] sofia_contact strange behavior
In-Reply-To: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>
References: <2160023e1003101252v7118d3b6mb6a99d199f6fc4bf@mail.gmail.com>
Message-ID: <191c3a031003101431u4ff81841nc19115c8f0bbaf46@mail.gmail.com>

if you pulled that from the user's xml you need to replace dialed_ext with
something else.
That is a special var only relevant as a user param.


On Wed, Mar 10, 2010 at 2:52 PM, RobertT <siniypin at gmail.com> wrote:

> Hello everybody!
>
> I am experiencing problems with sofia_contact command.
> When I dial sofia_contact external/1000@<domain> in fs cli I see correct
> registration data, provided the user 1000 is registered.
> But in the same time when I call ${sofia_conatct(external/${dialed_ext}@$${domain})}
> from dialplan as an argument to deflect application I recieve
> error\user_not_registered.
> What the hell?
>
> Best regards, RobertT.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ledoktre at meanie.us  Wed Mar 10 14:42:21 2010
From: ledoktre at meanie.us (Tim Streit)
Date: Wed, 10 Mar 2010 16:42:21 -0600
Subject: [Freeswitch-users] Skypiax- how to know which card to use?
In-Reply-To: <7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>
References: <4B7F0302.3060303@meanie.us>	<7b197bef1002191441u455e8d8an4eb03c460254c6d3@mail.gmail.com>
	<4B8569CB.6070804@meanie.us>	<7b197bef1002241207k4da85b36l5e2e9c8e14f4fe32@mail.gmail.com>
	<4B980E77.8080303@meanie.us>
	<7b197bef1003101420p29049831q1d061c664dbd5208@mail.gmail.com>
Message-ID: <4B98204C.2080601@meanie.us>

Giovanni,

Thanks for your reply.  I wondered if I somehow missed it !!

Giovanni Maruzzelli wrote:
> On Wed, Mar 10, 2010 at 10:26 PM, Tim Streit <ledoktre at meanie.us> wrote:
>
> Ciao Tim
>
> hehehe, you've missed the announce, so!
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/026987.html
>   



From jbrucehopkins at gmail.com  Wed Mar 10 15:05:00 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Wed, 10 Mar 2010 23:05:00 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003100400v555bae72ub6a70cb546029e1c@mail.gmail.com>
	<d03a10e01003101328u4aa33c79x4708f4177072ac10@mail.gmail.com>
	<d03a10e01003101404j71567ae5y2fa6abbf004bdbaf@mail.gmail.com>
Message-ID: <d03a10e01003101505w116f3a38pfe09a29ffdf498b6@mail.gmail.com>

OK - I have put a Wireshark trace across the router between FS and the
internet, and it will not surprise you to hear that when I tried ot call
from LAN-->WAN, FS was sending packets to the private IP of the remote phone
(behind its own NAT).

So how to sort this?

I tried:

1. Commenting these out in the internal profile

<!--all inbound reg will look in this domain for the users -->
    <!--<param name="force-register-domain" value="$${domain}"/>-->
    <!--force the domain in subscriptions to this value -->
    <!--<param name="force-subscription-domain" value="$${domain}"/>-->
    <!--all inbound reg will stored in the db using this domain -->
    <!--<param name="force-register-db-domain" value="$${domain}"/>-->

2. then creating a n antry in the directory refering to the same default
group of extensions as the default.xml, but with:
<domain name="dns.aname.for.FS">

and in the variables:

 <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

which I hoped from what I read in the Wiki might do the trick vis a vis
forceing FS to send the SIP packets to the public IP of the remote phone.

Alas it does not seem to work, and even phoning in from the WAS -->LAN now I
see USER_NOT_REGISTERED errors .  I get the same error in both directions.

Please please please could somebody put me out of my (your ;-) ) misery here
....?

What do I configure to tell FS to send outbound INVITE's to the public IP
the remote phone registered from behind, rather than its private NAT'ed ip?

many thanks in advance.
Bruce




On 10 March 2010 22:04, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Interestingly I *can* call outbound to the WAN a SIP client registered
> with FS via Fring.  I wonder if Fring have some additional nat traversal/SBC
> malarkey going on as they proxy the calls.
>
> I can't call outbound to an Eyebeam softphone registered directly wth FS
> though - although said Eyebeam is registered and can call inwards.
>
> I must have my nat handling configured wrongly.
>
> B
>
>
> On 10 March 2010 21:28, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hey Roly,
>>
>> Are you getting anywhere?  I'm still struggling, I'm afraid.
>>
>> I'm able to call from WAN --> LAN using the internal profile on port 5060,
>> but I still can't call out from LAN --> WAN so there is obviously something
>> really basic I'm still not getting.
>>
>> In sip_profiles/internal.xml I have:
>>
>>
>>  <param name="aggressive-nat-detection" value="true"/>
>> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>
>> <!-- the external_rtp_ip etc being defined in vars.xml -->
>>
>>
>> In this example I have left <param name="force-register-domain"
>> value="$${domain}"/> etc uncommented to force the registration to go to
>> default.xml in the dialplan.
>>
>> I thought maybe this was my problem and tried putting a separate file in
>> the directory to enable the WAN phone to register using the external DNS
>> name for the server as its domain, and uncommenting force-register-domain,
>> etc.  But then I got an error saying that the local (to Freeswitch) phone in
>> the LAN was not registered when I tried to make the call from the WAN.
>>
>> Any thoughts?
>>
>> cheers
>> Bruce
>>
>>
>>
>>
>>
>>
>>
>> On 10 March 2010 12:00, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>>
>>> Hi Roly,
>>>
>>> I hope to be able to have a go at this later today.  I'll let you know
>>> what happens.
>>>
>>> I have my FS box behind a hardware router/firewall, so I'm assuming I
>>> just forward SIP port 5060 from the WAN to the FS box and use internal
>>> profile, instead of using external profile with port 5080 forwarded as I
>>> have at the moment.  We'll see what happens.
>>>
>>> cheers
>>> Bruce
>>>
>>> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>>>
>>>> Ah I see... what? Please share and lead this blind man out the FS
>>>> wilderness!
>>>>
>>>> I don't understand...what happens to the external profile? Do you delete
>>>> it?
>>>> And how do you forward port 5060?
>>>>
>>>> ...and you thought you were a newbie!
>>>>
>>>> Any insight would be much appreciated...loving the journey.
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>>> Bruce
>>>> Hopkins
>>>> Sent: 09 March 2010 08:23 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>>>>
>>>> Ah I see.  I will try again using the internal profile and forwarding
>>>> port 5060.  Presumably still creating a directory entry to enable the
>>>> outside-facing domain to be used.
>>>>
>>>> Many thanks for your patient help of a newbie Brian.
>>>>
>>>> Cheers
>>>> Bruce
>>>>
>>>> Please excuse the brevity - sent from my mobile.
>>>>
>>>> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> > ok you aren't catching one key thing here.. you no longer need two
>>>> > profiles.
>>>> >
>>>> > /b
>>>> >
>>>> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>>>> >
>>>> >> Hi again,
>>>> >>
>>>> >> Thanks for the help so far.  Unfotrunately I must still be doing
>>>> >> something wrong here as I am still having difficulty, and still
>>>> >> have the same problem.
>>>> >>
>>>> >> I updated to build 16938 by means of "make current"
>>>> >>
>>>> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
>>>> >> to use 5080 instead of 5090.
>>>> >>
>>>> >> I got rid of the doublenat profile in sip_profiles, though I had to
>>>> >> retain an entry in the directory /usr/local/freeeswitch/conf/
>>>> >> directory/ext_dns.xml in order to give freeswitch the dns name of
>>>> >> the server as a domain for the remote softphone to register on.  I
>>>> >> left the group name in this entry the same as inthe default entry,
>>>> >> so that the remote phone could register on the same extension
>>>> >> numbers (100, etc) as in the default build.
>>>> >>
>>>> >> I still find that, if I initiate a call from the local (on same LAN
>>>> >> as freeswitch) phone to the remote phone, I get the message on the
>>>> >> CLI:
>>>> >>
>>>> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>>>> >> type [user] cause: [USER_NOT_REGISTERED]
>>>> >>
>>>> >> One possibly unrelated aside, I also found I needed to uncomment
>>>> >> <param name="aggressive-nat-detection" value="true"/> in
>>>> >> external.xml, otherwise in the case of a call initiated by the
>>>> >> remote phone being hung up by the local phone, freeswitch sent the
>>>> >> BYE to the private IP of the remote phone, rather than its public
>>>> >> ip - meaning that the remote phone didn't receive the BYE.
>>>> >>
>>>> >> Any further ideas where I am going wrong here please?
>>>> >>
>>>> >> thanks again in advance
>>>> >> Bruce
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> > users
>>>> > http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
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From anthony.minessale at gmail.com  Wed Mar 10 15:12:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 17:12:44 -0600
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
	<019801cac087$a9da3b90$fd8eb2b0$@com>
	<191c3a031003101229x1ddaa5b8k1d6a487e46580b1a@mail.gmail.com>
	<021b01cac093$c76d3930$5647ab90$@com>
	<7b197bef1003101423sfffb200ga206bcc5029d0a21@mail.gmail.com>
Message-ID: <191c3a031003101512u77744760h375c94f05d42a9fa@mail.gmail.com>

it means that it has detected a 150 microsecond offset.

It tries originally to sleep 1000 microseconds (1 millisecond) in a loop.
It times how long the sleep lasted and compares how long it really took vs
how long it was asked to sleep.
then it slowly decreases by 50 and tries again until it gets within 2
microseconds of the goal.

in your case of 150, it means if you subtract 150 from the total number of
microseconds from all the sleeps
they will be more accurate.



On Wed, Mar 10, 2010 at 4:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
> > OK, can you decode what this means exactly?  I?d be happy to write
> something
> > up to reference users to, but I don?t really know what it means.  It is a
> a
> > quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
> > 150, but is that good, bad, indifferent?  I?ve tried to google it, but
> > without really knowing what it is I am looking for, I am not getting good
> > results.
>
> This tells you the calculated average discrepancy of the timing is 150
> microseconds (millisecond/1000), that I believe is good. (hope this
> explanation is true ;) )
>
>
> >
> >
> >
> >
> >
> > 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating
> timer,
> > please wait...
> >
> > 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000
> Average:
> > 2001 Step: 50
> >
> > 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650 Average:
> > 1997 Step: 50
> >
> > 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200 Average:
> > 1038 Step: 50
> >
> > 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
> > 1018 Step: 50
> >
> > 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 998
> > Step: 50
> >
> > 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average:
> 1018
> > Step: 50
> >
> > 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000
> Average:
> > 2018 Step: 50
> >
> > 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950 Average:
> > 1998 Step: 50
> >
> > 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900 Average:
> > 1578 Step: 50
> >
> > 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850 Average:
> > 998 Step: 50
> >
> > 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of
> 150
> > calculated
> >
> > 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [CORE_SOFTTIMER_MODULE]
> >
> > 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
> > Timer 'soft'
> >
> > 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
> > Successfully Loaded [CORE_PCM_MODULE]
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Wed Mar 10 15:15:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Mar 2010 17:15:59 -0600
Subject: [Freeswitch-users] Tone Detection During Bridge
In-Reply-To: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>
References: <fb4ab1e41003101305r4bdf2faej34bb1c0acf23e09a@mail.gmail.com>
Message-ID: <191c3a031003101515r415299dckdea8d73369c0d89b@mail.gmail.com>

it should be running during the bridge as well.
once you start it, it attaches in the background and snoops all of your
audio.

Try turning up the debug level to see if it may be exiting.

"console loglevel debug"


On Wed, Mar 10, 2010 at 3:05 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Dear List,
>
> I have a requirement (due to poor quality Sipura products) to have
> freeswitch listen for a tone during a bridge (standard local extension
> dialplan segment). I have the required line:
>
> <action appication="tone_detect" data="hangup 400 r 0 hangup
> 'normal_clearing' 1"/>
>
> But I need to have this run _during_ the bridge (to listen for the tone
> coming from the broken SPA3k and to break the bridge). Does anyone have any
> idea how this may be achieved? There is no such documentation on the wiki.
> This works perfectly if the bridge was not successful, and so will detect
> tones during voicemail messaging.
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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From ben at langfeld.co.uk  Wed Mar 10 15:57:02 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 10 Mar 2010 23:57:02 +0000
Subject: [Freeswitch-users] Tone Detection During Bridge (Anthony
	Minessale)
Message-ID: <fb4ab1e41003101557r733f148o72f87276f11d87b3@mail.gmail.com>

Cheers Anthony, I don't know how I was overcomplicating that so much, but it
works fine now. Thanks!

Regards,
Ben Langfeld

Wave > Email


On Wed, Mar 10, 2010 at 11:16 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Kernel timer warning with CentOS 5.4 (Anthony Minessale)
>   2. Re: Tone Detection During Bridge (Anthony Minessale)
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 10 Mar 2010 17:12:44 -0600
> Subject: Re: [Freeswitch-users] Kernel timer warning with CentOS 5.4
> it means that it has detected a 150 microsecond offset.
>
> It tries originally to sleep 1000 microseconds (1 millisecond) in a loop.
> It times how long the sleep lasted and compares how long it really took vs
> how long it was asked to sleep.
> then it slowly decreases by 50 and tries again until it gets within 2
> microseconds of the goal.
>
> in your case of 150, it means if you subtract 150 from the total number of
> microseconds from all the sleeps
> they will be more accurate.
>
>
>
> On Wed, Mar 10, 2010 at 4:23 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:
>
>> On Wed, Mar 10, 2010 at 9:53 PM, Peder <peder at networkoblivion.com> wrote:
>> > OK, can you decode what this means exactly?  I?d be happy to write
>> something
>> > up to reference users to, but I don?t really know what it means.  It is
>> a a
>> > quad-core 2.8GHz running CentOS 5.3 64-bit.  It looks like it calculated
>> > 150, but is that good, bad, indifferent?  I?ve tried to google it, but
>> > without really knowing what it is I am looking for, I am not getting
>> good
>> > results.
>>
>> This tells you the calculated average discrepancy of the timing is 150
>> microseconds (millisecond/1000), that I believe is good. (hope this
>> explanation is true ;) )
>>
>>
>> >
>> >
>> >
>> >
>> >
>> > 2010-03-10 14:48:37.960041 [CONSOLE] switch_time.c:1021 Calibrating
>> timer,
>> > please wait...
>> >
>> > 2010-03-10 14:48:38.060174 [CONSOLE] switch_time.c:223 Test: 1000
>> Average:
>> > 2001 Step: 50
>> >
>> > 2010-03-10 14:48:38.160169 [CONSOLE] switch_time.c:223 Test: 950
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.260164 [CONSOLE] switch_time.c:223 Test: 900
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.360159 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.460155 [CONSOLE] switch_time.c:223 Test: 800
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.560149 [CONSOLE] switch_time.c:223 Test: 750
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.660144 [CONSOLE] switch_time.c:223 Test: 700
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.760139 [CONSOLE] switch_time.c:223 Test: 650
>> Average:
>> > 1997 Step: 50
>> >
>> > 2010-03-10 14:48:38.860134 [CONSOLE] switch_time.c:223 Test: 600
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:38.960132 [CONSOLE] switch_time.c:223 Test: 550
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.060124 [CONSOLE] switch_time.c:223 Test: 500
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.160119 [CONSOLE] switch_time.c:223 Test: 450
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.260114 [CONSOLE] switch_time.c:223 Test: 400
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:39.311111 [CONSOLE] switch_time.c:223 Test: 350
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:39.361109 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.411106 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.461103 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.511101 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.561099 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.612096 [CONSOLE] switch_time.c:223 Test: 300
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:39.662092 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.712091 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.762088 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.812086 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.862083 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.912082 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:39.962077 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.012076 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.063074 [CONSOLE] switch_time.c:223 Test: 250
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:40.113071 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.163068 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.213066 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.263064 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.313061 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.363059 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.413056 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.465054 [CONSOLE] switch_time.c:223 Test: 200
>> Average:
>> > 1038 Step: 50
>> >
>> > 2010-03-10 14:48:40.515051 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.565049 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.615046 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.665043 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.715041 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.765039 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.815036 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.866032 [CONSOLE] switch_time.c:223 Test: 150
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:40.916032 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:40.966028 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.016026 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.066022 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.116021 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.166017 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.216016 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.266013 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:41.317011 [CONSOLE] switch_time.c:223 Test: 100
>> Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:41.367008 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.417006 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.467004 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.517001 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.566999 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.616996 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.666993 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.716991 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.767989 [CONSOLE] switch_time.c:223 Test: 50 Average:
>> > 1018 Step: 50
>> >
>> > 2010-03-10 14:48:41.817985 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.867984 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.917980 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:41.967979 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.017975 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.067974 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.117970 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 998
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.168969 [CONSOLE] switch_time.c:223 Test: 0 Average:
>> 1018
>> > Step: 50
>> >
>> > 2010-03-10 14:48:42.269962 [CONSOLE] switch_time.c:223 Test: 1000
>> Average:
>> > 2018 Step: 50
>> >
>> > 2010-03-10 14:48:42.369958 [CONSOLE] switch_time.c:223 Test: 950
>> Average:
>> > 1998 Step: 50
>> >
>> > 2010-03-10 14:48:42.448958 [CONSOLE] switch_time.c:223 Test: 900
>> Average:
>> > 1578 Step: 50
>> >
>> > 2010-03-10 14:48:42.498954 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.548951 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.598949 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.648946 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.698945 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.748942 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.798941 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.848936 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.898935 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.948931 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.998931 [CONSOLE] switch_time.c:223 Test: 850
>> Average:
>> > 998 Step: 50
>> >
>> > 2010-03-10 14:48:42.998963 [CONSOLE] switch_time.c:274 Timer offset of
>> 150
>> > calculated
>> >
>> > 2010-03-10 14:48:42.998994 [CONSOLE] switch_loadable_module.c:900
>> > Successfully Loaded [CORE_SOFTTIMER_MODULE]
>> >
>> > 2010-03-10 14:48:42.999040 [NOTICE] switch_loadable_module.c:230 Adding
>> > Timer 'soft'
>> >
>> > 2010-03-10 14:48:42.999234 [CONSOLE] switch_loadable_module.c:900
>> > Successfully Loaded [CORE_PCM_MODULE]
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 10 Mar 2010 17:15:59 -0600
> Subject: Re: [Freeswitch-users] Tone Detection During Bridge
> it should be running during the bridge as well.
> once you start it, it attaches in the background and snoops all of your
> audio.
>
> Try turning up the debug level to see if it may be exiting.
>
> "console loglevel debug"
>
>
> On Wed, Mar 10, 2010 at 3:05 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Dear List,
>>
>> I have a requirement (due to poor quality Sipura products) to have
>> freeswitch listen for a tone during a bridge (standard local extension
>> dialplan segment). I have the required line:
>>
>> <action appication="tone_detect" data="hangup 400 r 0 hangup
>> 'normal_clearing' 1"/>
>>
>> But I need to have this run _during_ the bridge (to listen for the tone
>> coming from the broken SPA3k and to break the bridge). Does anyone have any
>> idea how this may be achieved? There is no such documentation on the wiki.
>> This works perfectly if the bridge was not successful, and so will detect
>> tones during voicemail messaging.
>>
>> Regards,
>> Ben Langfeld
>>
>> Wave > Email
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Wed Mar 10 16:41:55 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 11 Mar 2010 03:41:55 +0300
Subject: [Freeswitch-users] Switch with External SoftPhone
Message-ID: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>

Hello.

What are the ways connect external users to the fs there? In such a scheme:
 fs with real ip <-> internet <-> nat <-> sip-phone


I read the topic http://wiki.freeswitch.org/wiki/External_profile #
Switch_with_External_SoftPhone

I did not understand several things:
- That should be in exteranl5090.xml? Whole profile external.xml c changes
in paragraphs 2,3?
- Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in
xxxx? External UA IP? Domain?" I have the same question. that x.x.x.x should
be?

But this seems not quite what I need. Must be external to the subscriber is
practically no different from internal.

-- 
Best regards, Vladimir Elizarov
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From lloyd.aloysius at gmail.com  Wed Mar 10 16:59:06 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 10 Mar 2010 19:59:06 -0500
Subject: [Freeswitch-users] Multiple Registrations - randomly only one
	device ring
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>
References: <8a19bf2e1003092353w32701aaes47b034f2abfb71f5@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005C5B@VMBX113.ihostexchange.net>
	<8a19bf2e1003101352w31f92510ld613602fd1b04f64@mail.gmail.com>
	<7454A296C7EDE34EA57199FAA401E2F11C64005D81@VMBX113.ihostexchange.net>
Message-ID: <8a19bf2e1003101659x29ba2620k888b4a4fc2a3b546@mail.gmail.com>

Thank you for the help.



On Wed, Mar 10, 2010 at 5:26 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Hi Lloyd,
>
>
>
> It depends on which profile your phones are registering with.  I suspect
> internal.xml will work.
>
>
>
> Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Wednesday, March 10, 2010 1:52 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Multiple Registrations - randomly only
> one device ring
>
>
>
> Chris, Thank you.
>
>
>
> I find the following information from wiki.This is say sofia.conf. I think
> sofia.conf.xml
>
>
>  Multiple Registrations
>
> Call one extension and ring several phones
>
> You must enable multiple registrations in sofia.conf
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> ----
>
> Which file I should enable internal.xml or sofia.conf.xml
>
> Thanks
>
> Lloyd
>
>
>
>
>
> On Wed, Mar 10, 2010 at 12:16 PM, Chris Fowler <chris at fowler.cc> wrote:
>
> Hi Lloyd,
>
>
>
> You need to use sofia_contact in your bridge command in the dialplan:
>
>
>
> e.g.
>
> *<anti-action application="bridge"
> data="${sofia_contact(${dialed_extension}@${domain_name})}"/>*
>
> * *
>
> This will cause FS to try all registrations simultaneously.
>
>
>
> Cheers, Chris.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Aloysius
> Lloyd
> *Sent:* Tuesday, March 09, 2010 11:53 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Subject:* [Freeswitch-users] Multiple Registrations - randomly only one
> device ring
>
>
>
> Hi All,
>
>
>
> I enable the multiple registrations in sip_profiles/internal.xml.
>
>
>
> <param name="multiple-registrations" value="true"/>
>
>
>
> Then I restart the freeswitch.
>
>
>
> I use Aastra 9133i and eyebeam register for same extension.
>
>
>
> But randomly only one device ring. Do I need to enable any other settings?
>
>
>
>
> Thanks,
> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lakindia89 at gmail.com  Wed Mar 10 20:07:06 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 11 Mar 2010 09:37:06 +0530
Subject: [Freeswitch-users] smg_prid not bridging the call
In-Reply-To: <F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>
References: <6d15d07f1002270033n71d8ac85u1895b05f75540e63@mail.gmail.com>
	<7d79b3931003042011v154fd1d4g136f4b4e2f6800a@mail.gmail.com>
	<F7E94B21-372A-4B0E-9AE5-3CB2CD08978F@jerris.com>
Message-ID: <7d79b3931003102007n728e2c3dudf603aafedb18b6b@mail.gmail.com>

My switch type is EuroISDN, and I used Q931 as the dialect.


On Wed, Mar 10, 2010 at 3:31 PM, Michael Jerris <mike at jerris.com> wrote:

> What pri dialect is this?
>
> On Mar 4, 2010, at 11:11 PM, lakshmanan ganapathy wrote:
>
> Ok. This is to say how the problem got solved.
> Need openzap 1047 or above version.
> Need wanpipe-3.5.8.6.smg_pri-v1.63.tgz
> My telco is not accepting the display IE.
> Finally setting disable_display_ie=yes in smg_pri.conf solved the issue.
>
> Thanks for all.
>
> On Sat, Feb 27, 2010 at 2:03 PM, Tomasz Augustyn <tomek.augustyn at gmail.com
> > wrote:
>
>> Hello,
>>
>> I had similar problem and I think it is more a problem between Sangoma
>> card and your E1 provider than with freeswitch.
>>
>> In my case it was necessary to set "origination_caller_id_number" to one
>> of the telephone numbers linked to my E1 line. In other case the calls were
>> rejected with "invalid information element" error.
>>
>> You can try Sangoma's support they are very helpful.
>>
>> Tomasz Augustyn
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lvatland at gmail.com  Wed Mar 10 07:31:28 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Wed, 10 Mar 2010 09:31:28 -0600
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
Message-ID: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?


Thanks for any help.


Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com



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From lvatland at gmail.com  Wed Mar 10 14:38:19 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Wed, 10 Mar 2010 16:38:19 -0600
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
Message-ID: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?


Thanks for any help.


Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com

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From arliyardo at gmail.com  Wed Mar 10 17:25:50 2010
From: arliyardo at gmail.com (arliyardo)
Date: Thu, 11 Mar 2010 09:25:50 +0800
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls between two NAT?
Message-ID: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>

my scenario like this:

PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET ->
NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
First, I have forwarded sip port and rtp port.
I have try servaral configurations in
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and
without luck!

PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2,
include sip signaling and rtp connection ?
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From helmut.kuper at ewetel.de  Thu Mar 11 00:34:41 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 11 Mar 2010 09:34:41 +0100
Subject: [Freeswitch-users] openzap  Q.SIG and Q.931
Message-ID: <4B98AB21.5010102@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

maybe you know that I run a long time test with Stefan Knoblich's q931
pri-stack for openzap. In fact, I use is in my production system since a
year or so. I have to say that it is very stable and I'm happy with it -
ususally ... but:

I wonder if anyone of you is working on a Q.SIG Basic-Call Support
(ECMA-143) for openzap. I try to connect to an Avaya system using Q.SIG
to be a part of the internal domain. My main targets are to send and
receive Caller's and Callee's name and to remove a leading '0' (which is
added by AVAYA because of q931) for internal calls.

regards
helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLmKsh4tZeNddg3dwRAjaNAJ4pTdxYyQizHPDXP9gjjUyABJIuxwCcDnNj
qlCG9cdIfjDLlQE3bQfxluw=
=L1tJ
-----END PGP SIGNATURE-----



From mike at jerris.com  Thu Mar 11 01:09:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Mar 2010 04:09:03 -0500
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
In-Reply-To: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
References: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
Message-ID: <DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>

I think I fixed this about a week ago.

On Mar 10, 2010, at 10:31 AM, Larry Vatland wrote:

> I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working?
> 




From jbrucehopkins at gmail.com  Thu Mar 11 02:03:39 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Thu, 11 Mar 2010 10:03:39 +0000
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls 	between two NAT?
In-Reply-To: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
References: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
Message-ID: <d03a10e01003110203t29ebc270l73594f28463b1e96@mail.gmail.com>

Hello arliyardo,

I'm new to Freeswitch, but looking at the posts on the mailing list it
appears a few of us - you, me and Vladimir Elizarov, have been grappling
with he same thing recently - getting remote phones to work through NAT.

If you look at my thread you will see I have got some of the way, and can
call incoming now but still have difficulty with outgoing calls.  Perhaps
you might have a look at where I have got to and see if you can think of the
final bit of the picture?

I gather the developers are in the process of updating the Wiki pages on
NAT.  I'm hoping they will be able to find time to continuw with this.  It
must be a laborious and time consuming task keeping the documentation up to
date.

bests
Bruce

On 11 March 2010 01:25, arliyardo <arliyardo at gmail.com> wrote:

> my scenario like this:
>
> PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET -> NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
> First, I have forwarded sip port and rtp port.
> I have try servaral configurations in http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and without luck!
>
> PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2, include sip signaling and rtp connection ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Mar 11 06:10:37 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Mar 2010 08:10:37 -0600
Subject: [Freeswitch-users] The demo extension mad_boss_intercom problem
In-Reply-To: <DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>
References: <7F24C234-F46B-40AE-BB6D-293D6322F903@gmail.com>
	<DC28D63F-4103-4ACB-A874-E46B3F8512B5@jerris.com>
Message-ID: <6A9394D0-14C4-489C-8018-B79D1E741A8F@freeswitch.org>

I don't think... I know you fixed it about a week ago :P

/b

On Mar 11, 2010, at 3:09 AM, Michael Jerris wrote:

> I think I fixed this about a week ago.




From phunk0000 at hotmail.com  Thu Mar 11 07:44:02 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 11 Mar 2010 10:44:02 -0500
Subject: [Freeswitch-users] can't get programming language modules to load
Message-ID: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>

Hey list-  I have make mod_whatever-installed all the individual mod I want
and have been able to make all but the following work.  I have followed wiki
directions for odbc and have odbc working.  Any help in getting these
modules to load would be great.  Thanks

 

**/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
mod_spidermonkey_core_db_module_interface**

2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so

**/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
mod_spidermonkey_curl_module_interface**

2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so

**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
mod_spidermonkey_odbc_module_interface**

2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so

**/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
mod_spidermonkey_socket_module_interface**

2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so

**/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
mod_spidermonkey_teletone_module_interface**

2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_perl.so

**/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
such file or directory**

2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_python.so

**/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
No such file or directory**

2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so

**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such file or directory**

 

Thanks

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From brian at freeswitch.org  Thu Mar 11 07:51:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Mar 2010 09:51:06 -0600
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
Message-ID: <42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>

Those are sub modules of spidermonkey not FreeSWITCH.... Please see the spidermonkey.conf.xml

And the others are not compiled.

/b

On Mar 11, 2010, at 9:44 AM, Todd wrote:

> Hey list-  I have make mod_whatever-installed all the individual mod I want and have been able to make all but the following work.  I have followed wiki directions for odbc and have odbc working.  Any help in getting these modules to load would be great.  Thanks
>  
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol: mod_spidermonkey_core_db_module_interface**
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface**
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: mod_spidermonkey_odbc_module_interface**
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol: mod_spidermonkey_socket_module_interface**
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol: mod_spidermonkey_teletone_module_interface**
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_perl.so
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No such file or directory**
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_python.so
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: No such file or directory**
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No such file or directory**
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From phunk0000 at hotmail.com  Thu Mar 11 08:31:24 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 11 Mar 2010 11:31:24 -0500
Subject: [Freeswitch-users] can't get programming language modules
	to	load
In-Reply-To: <42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
Message-ID: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>

Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
still get those same errors.  I have also attempted to 'make
mod_java-install' for all the uncompiled ones and it runs through a bunch of
stuff on the screen but ends with 'make[1]: *** [mod_java.lo] error 1' same
with python and lua.  any info on how to properly compile the others and get
the spidermokney subs to work would be great.  thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 11, 2010 10:51 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] can't get programming language modules to
load

 

Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
spidermonkey.conf.xml

 

And the others are not compiled.

 

/b

 

On Mar 11, 2010, at 9:44 AM, Todd wrote:





Hey list-  I have make mod_whatever-installed all the individual mod I want
and have been able to make all but the following work.  I have followed wiki
directions for odbc and have odbc working.  Any help in getting these
modules to load would be great.  Thanks

 

**/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
mod_spidermonkey_core_db_module_interface**

2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so

**/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
mod_spidermonkey_curl_module_interface**

2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so

**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
mod_spidermonkey_odbc_module_interface**

2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so

**/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
mod_spidermonkey_socket_module_interface**

2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so

**/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
mod_spidermonkey_teletone_module_interface**

2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_perl.so

**/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
such file or directory**

2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_python.so

**/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
No such file or directory**

2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading
module /usr/local/freeswitch/mod/mod_java.so

**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such file or directory**

 

Thanks

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
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From tculjaga at gmail.com  Thu Mar 11 09:15:16 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Mar 2010 18:15:16 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>
	<191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
Message-ID: <65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>

On Wed, Mar 10, 2010 at 6:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We have a mod_fax for t30 and work on t38 is underway.
> So we already acknowledge that I guess.
>
>
>
anything ready for testing ?
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From tculjaga at gmail.com  Thu Mar 11 09:19:01 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Mar 2010 18:19:01 +0100
Subject: [Freeswitch-users] Kernel timer warning with CentOS 5.4
In-Reply-To: <191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
References: <d03a10e01002280737l61e1db84xa8388a73e932d460@mail.gmail.com>
	<d03a10e01002281003j669b6e75t31db4dc351f38ece@mail.gmail.com>
	<191c3a031003010847y5036817ekb19379e646dd8cb@mail.gmail.com>
	<d03a10e01003080525p572ff274nce70fdb97f1197cf@mail.gmail.com>
	<65d96fc81003100142m129c6b66w229bd1e519e8aea7@mail.gmail.com>
	<191c3a031003100825s7d05562cna52d39265bfcf54@mail.gmail.com>
Message-ID: <65d96fc81003110919i7a9bcc87xd92edeab6598fc0b@mail.gmail.com>

On Wed, Mar 10, 2010 at 5:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> FS now tries to determine how accurate the timing is by doing some tests
> with 1ms sleeps.
> Most 64 bit CentOS box I have seen can easily calibrate.
> If you are having that error alot but the calls still sound ok, it's safe
> to ignore it.
> you can also start FS with -nocal to skip this test.
>
> Also make sure you do not have any other cpu-intensive application on your
> box possibly competing for resources.
>
>
Right, this happens on 32bit Centos (at least in my case) only... on 64bit
is is ok ... and no complains at all :)

misery solved for me.... thx.


T.
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From ken at ukgb.net  Thu Mar 11 09:34:08 2010
From: ken at ukgb.net (Ken Gillett)
Date: Thu, 11 Mar 2010 17:34:08 +0000
Subject: [Freeswitch-users] PSTN connection
Message-ID: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>

If I am running FreeSwitch on a Mac, would I be able to use Apple's USB modem as a means to connect to the PSTN and utilise this line to place and receive calls?



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From freeswitch at cartissolutions.com  Thu Mar 11 10:35:30 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 11 Mar 2010 12:35:30 -0600
Subject: [Freeswitch-users] FS no frok mode
In-Reply-To: <8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>
References: <8ccbff061003092110y5baddaecmf30c7015996e519b@mail.gmail.com>	<5A67C661-6E80-423F-81E9-B56B0E4299A0@jerris.com>
	<8ccbff061003100212t414baedat9cf84047cef05d02@mail.gmail.com>
Message-ID: <4B9937F2.2010807@cartissolutions.com>


On 03/10/2010 04:12 AM, Dome Charoenyost wrote:
> 2010/3/10 Michael Jerris<mike at jerris.com>:
>    
>> -nf has nothing to do with the performance, it has to do if the command
>> blocks or not when you run it.  I am unsure what you mean about sql in this
>> context, it is totally unrelated. What is a cash db?
>>      
> sorry i mean switch_cache_db_connection :)
> i have problem about db connection. now switch_cache_db_connection use
> 1 connection for 1 thread
> if i use mod_limit with odbc my server can't handle more than 100 connection.
> So i'll try FS in no fork. but i want to know other effect.
>
> BG
>
> Dome C.
>    

I think you misunderstand what it is that the -nf option does.  It will 
have no bearing on how switch's odbc caching mechanisms work.  For 
example, I run freeswitch under a supervising daemon (djb's daemontools 
in this case) with -nf and -nc options.  This prevents freeswitch from 
spawning the console, and it also prevents freeswitch from forking to 
the background.  In otherwords, it keeps freeswitch from becoming a 
background daemon.


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com





From testeador01 at gmail.com  Thu Mar 11 11:24:25 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 11 Mar 2010 14:24:25 -0500
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
	<blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
Message-ID: <c29b930b1003111124m10b85e08t9fae1067a54b4d9b@mail.gmail.com>

Hello Todd,

I know this can be a little complicated at first but if you research the
wiki and google you can succeed,

You didn't need to just "uncomment" them, you need to install them so they
can actually run...

to install spidermonkey odbc, there are actually some packages that have to
be installed on your linux box, you can find them on the wiki for SPIDERMONKEY
ODBC <http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc> not just the
one for odbc, in that wiki, they tell you what you need to install.

to install mod java you need to run the configure option that is indicated
in the mod_java wiki as well...

if you still get errors after that, please post back, but post the whole
error message, saying "error 1" doesn't help us understand what are you
missing.

Good luck with this :)


2010/3/11 Todd <phunk0000 at hotmail.com>

>  Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
> still get those same errors.  I have also attempted to ?make
> mod_java-install? for all the uncompiled ones and it runs through a bunch of
> stuff on the screen but ends with ?make[1]: *** [mod_java.lo] error 1? same
> with python and lua?  any info on how to properly compile the others and get
> the spidermokney subs to work would be great.  thanks
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Thursday, March 11, 2010 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] can't get programming language modules
> to load
>
>
>
> Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
> spidermonkey.conf.xml
>
>
>
> And the others are not compiled.
>
>
>
> /b
>
>
>
> On Mar 11, 2010, at 9:44 AM, Todd wrote:
>
>
>
>   Hey list-  I have make mod_whatever-installed all the individual mod I
> want and have been able to make all but the following work.  I have followed
> wiki directions for odbc and have odbc working.  Any help in getting these
> modules to load would be great.  Thanks
>
>
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
> mod_spidermonkey_core_db_module_interface**
>
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
> mod_spidermonkey_curl_module_interface**
>
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
> mod_spidermonkey_odbc_module_interface**
>
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
> mod_spidermonkey_socket_module_interface**
>
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
> mod_spidermonkey_teletone_module_interface**
>
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_perl.so
>
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
> such file or directory**
>
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_python.so
>
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
> No such file or directory**
>
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so
>
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such file or directory**
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
> 02:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testeador01 at gmail.com  Thu Mar 11 11:24:25 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 11 Mar 2010 14:24:25 -0500
Subject: [Freeswitch-users] can't get programming language modules to
	load
In-Reply-To: <blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
References: <BLU110-DS17C58A0966E6D64C8DE1B2A7320@phx.gbl>
	<42D518A4-A111-42E0-A33B-F1AA7BD69ADB@freeswitch.org>
	<blu110-ds185222B6FDCD8FD987FFB8A7320@phx.gbl>
Message-ID: <c29b930b1003111124m10b85e08t9fae1067a54b4d9b@mail.gmail.com>

Hello Todd,

I know this can be a little complicated at first but if you research the
wiki and google you can succeed,

You didn't need to just "uncomment" them, you need to install them so they
can actually run...

to install spidermonkey odbc, there are actually some packages that have to
be installed on your linux box, you can find them on the wiki for SPIDERMONKEY
ODBC <http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc> not just the
one for odbc, in that wiki, they tell you what you need to install.

to install mod java you need to run the configure option that is indicated
in the mod_java wiki as well...

if you still get errors after that, please post back, but post the whole
error message, saying "error 1" doesn't help us understand what are you
missing.

Good luck with this :)


2010/3/11 Todd <phunk0000 at hotmail.com>

>  Yes, I uncommented everything in the spidermonkey.conf.xml already, and I
> still get those same errors.  I have also attempted to ?make
> mod_java-install? for all the uncompiled ones and it runs through a bunch of
> stuff on the screen but ends with ?make[1]: *** [mod_java.lo] error 1? same
> with python and lua?  any info on how to properly compile the others and get
> the spidermokney subs to work would be great.  thanks
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Thursday, March 11, 2010 10:51 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] can't get programming language modules
> to load
>
>
>
> Those are sub modules of spidermonkey not FreeSWITCH.... Please see the
> spidermonkey.conf.xml
>
>
>
> And the others are not compiled.
>
>
>
> /b
>
>
>
> On Mar 11, 2010, at 9:44 AM, Todd wrote:
>
>
>
>   Hey list-  I have make mod_whatever-installed all the individual mod I
> want and have been able to make all but the following work.  I have followed
> wiki directions for odbc and have odbc working.  Any help in getting these
> modules to load would be great.  Thanks
>
>
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol:
> mod_spidermonkey_core_db_module_interface**
>
> 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol:
> mod_spidermonkey_curl_module_interface**
>
> 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol:
> mod_spidermonkey_odbc_module_interface**
>
> 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol:
> mod_spidermonkey_socket_module_interface**
>
> 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>
> **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol:
> mod_spidermonkey_teletone_module_interface**
>
> 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_perl.so
>
> **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No
> such file or directory**
>
> 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_python.so
>
> **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file:
> No such file or directory**
>
> 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error
> Loading module /usr/local/freeswitch/mod/mod_java.so
>
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such file or directory**
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> No virus found in this incoming message.
> Checked by AVG - www.avg.com
> Version: 9.0.733 / Virus Database: 271.1.1/2736 - Release Date: 03/11/10
> 02:33:00
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:37:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:37:26 -0800
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
Message-ID: <87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>

How are your internal users registering to your FS? What's your FS setup? Do
you have more than one NIC?
-MC

On Wed, Mar 10, 2010 at 4:41 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Hello.
>
> What are the ways connect external users to the fs there? In such a
> scheme:
>  fs with real ip <-> internet <-> nat <-> sip-phone
>
>
> I read the topic http://wiki.freeswitch.org/wiki/External_profile #
> Switch_with_External_SoftPhone
>
> I did not understand several things:
> - That should be in exteranl5090.xml? Whole profile external.xml c changes
> in paragraphs 2,3?
> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in
> xxxx? External UA IP? Domain?" I have the same question. that x.x.x.x
> should be?
>
> But this seems not quite what I need. Must be external to the subscriber
> is practically no different from internal.
>
> --
> Best regards, Vladimir Elizarov
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:39:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:39:16 -0800
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
Message-ID: <87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>

On Wed, Mar 10, 2010 at 2:38 PM, Larry Vatland <lvatland at gmail.com> wrote:

> I'm new to freeswitch and just exploring the default dialplan. I have not
> been able to get the mad_boss_intercom example to work correctly, it pages
> fine but you should be able to press *2 to start a two-way conversation.
> Whats happening is the * toggles the phones mute. How do I set up the
> conference to pass the *2 off to the bind_meta_app to get a page to intercom
> system working?
>
> What phone are you using? It sounds like maybe the phone itself is
intercepting the * and not sending it along to FreeSWITCH.
-MC
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From msc at freeswitch.org  Thu Mar 11 12:45:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:45:40 -0800
Subject: [Freeswitch-users] No UPNP/NAT-PMP,
	no STUN! How can i make calls 	between two NAT?
In-Reply-To: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
References: <90dfb0f81003101725t396f3c40vdea6cbabcee7a2a0@mail.gmail.com>
Message-ID: <87f2f3b91003111245g2b4833a7re68ca87ca1d09353@mail.gmail.com>

On Wed, Mar 10, 2010 at 5:25 PM, arliyardo <arliyardo at gmail.com> wrote:

> my scenario like this:
>
> PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET -> NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100)
> First, I have forwarded sip port and rtp port.
> I have try servaral configurations in http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and without luck!
>
> PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2, include sip signaling and rtp connection ?
>
>
> The key here is that you have two devices that can totally mess with your
SIP and RTP: NAT1 and NAT2. What are these two devices? Secondly, what kind
of phone is PHONE2? Some phones work better than others. Third, have you
captured the SIP traffic behind each NAT device to see what's going on? You
won't be able to fix the problem until you actually find out what the
problem is and where it is happening. For the record, I've done this kind of
thing using X-Lite and Snom 300's as the PHONE2 in your example, so I know
it is possible. The trick is making sure that your NAT devices have their
SIP ALG's turned off. Check this page to see if you have one of these
devices:
http://wiki.freeswitch.org/wiki/ALG

-MC
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From pjintheusa at gmail.com  Thu Mar 11 12:48:42 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 11 Mar 2010 15:48:42 -0500
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
Message-ID: <367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>

Larry - also see the replies to your previous post on this subject - which I
would paraphrase as "make sure the FS version you are using to test is the
latest"

On Thu, Mar 11, 2010 at 3:39 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Mar 10, 2010 at 2:38 PM, Larry Vatland <lvatland at gmail.com> wrote:
>
>> I'm new to freeswitch and just exploring the default dialplan. I have not
>> been able to get the mad_boss_intercom example to work correctly, it pages
>> fine but you should be able to press *2 to start a two-way conversation.
>> Whats happening is the * toggles the phones mute. How do I set up the
>> conference to pass the *2 off to the bind_meta_app to get a page to intercom
>> system working?
>>
>> What phone are you using? It sounds like maybe the phone itself is
> intercepting the * and not sending it along to FreeSWITCH.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 11 12:56:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 11 Mar 2010 12:56:45 -0800
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
	<367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
Message-ID: <87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>

On Thu, Mar 11, 2010 at 12:48 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Larry - also see the replies to your previous post on this subject - which
> I would paraphrase as "make sure the FS version you are using to test is the
> latest"
>
Or, in other words, go to your FreeSWITCH source directory and type "make
current" and then go get a cup of coffee. When it is done restart FreeSWITCH
and test again...
-MC
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From srinivas.ksvreddy at gmail.com  Thu Mar 11 13:00:54 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Thu, 11 Mar 2010 16:00:54 -0500
Subject: [Freeswitch-users] more than one condition default.xml
Message-ID: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>

Hi,

can i have more than one conditions in externsion tag in
dialplan\default.xml.



-- 
Srinivasula Reddy K
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From anthony.minessale at gmail.com  Thu Mar 11 13:09:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Mar 2010 15:09:05 -0600
Subject: [Freeswitch-users] more than one condition default.xml
In-Reply-To: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
References: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
Message-ID: <191c3a031003111309v6d1dcceek41516267e6f3c9f1@mail.gmail.com>

yes but make sure they are stacked not nested (see wiki)

On Thu, Mar 11, 2010 at 3:00 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

>
> Hi,
>
> can i have more than one conditions in externsion tag in
> dialplan\default.xml.
>
>
>
> --
> Srinivasula Reddy K
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch at cartissolutions.com  Thu Mar 11 13:25:18 2010
From: freeswitch at cartissolutions.com (Yossi Neiman)
Date: Thu, 11 Mar 2010 15:25:18 -0600
Subject: [Freeswitch-users] more than one condition default.xml
In-Reply-To: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
References: <f8af5741003111300p3cc93c63iec8006cbdea75046@mail.gmail.com>
Message-ID: <4B995FBE.4090506@cartissolutions.com>

On 03/11/2010 03:00 PM, srinivasula reddy wrote:
>
> Hi,
> can i have more than one conditions in externsion tag in 
> dialplan\default.xml.
>
> -- 
> Srinivasula Reddy K
>

You can do the following:

<extension name="foo">
<condition ... />
<condition ... />
<condition ... >
<action ... />
<action ... />
<anti-action ... />
</condition>
<condition ... >
<action ... />
<anti-action ... />
</condition>
</extension>

But you can not have nested conditions.


Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com





From xengelpublicx at gmail.com  Thu Mar 11 13:25:52 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Fri, 12 Mar 2010 00:25:52 +0300
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
	<87f2f3b91003111237i643e444fx5ecb1bfff59adfbc@mail.gmail.com>
Message-ID: <eb0385731003111325j50686b27t83825ed89856fcfa@mail.gmail.com>

Internal users connects via internal profile.?FS has two interfaces lan and wan.
scheme:
wan(external real ip)<-fs->lan (local ip)-internal users

I need to have external users with the same capabilities as the internal users.

On Thu, Mar 11, 2010 at 11:37 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> How are your internal users registering to your FS? What's your FS setup? Do you have more than one NIC?
> -MC
>
> On Wed, Mar 10, 2010 at 4:41 PM, Vladimir Elizarov <xengelpublicx at gmail.com> wrote:
>>
>> Hello.
>>
>> What are the ways connect external users to the fs there??In such a scheme:
>> ?fs with real ip <-> internet <-> nat <-> sip-phone
>>
>>
>> I read the topic http://wiki.freeswitch.org/wiki/External_profile # Switch_with_External_SoftPhone
>>
>> I did not understand several things:
>> - That should be in exteranl5090.xml??Whole profile external.xml c changes in paragraphs 2,3?
>> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes in xxxx? External UA IP? Domain?"?I have the same question.?that x.x.x.x should be?
>>
>> But this seems not quite what I need.?Must be external to the subscriber is practically no different from internal.
>>
>> --
>> Best regards, Vladimir Elizarov
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Best regards, Vladimir Elizarov



From lvatland at gmail.com  Thu Mar 11 13:33:47 2010
From: lvatland at gmail.com (Larry Vatland)
Date: Thu, 11 Mar 2010 15:33:47 -0600
Subject: [Freeswitch-users] The demo xml dialplan extension
	mad_boss_intercom problem
In-Reply-To: <87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>
References: <A4EADFB1-3A53-4DCC-BA80-18BBC7E85578@gmail.com>
	<87f2f3b91003111239h1f07d7f7va4ffc9628cc217e7@mail.gmail.com>
	<367751821003111248k508db9dbn2c7a4d73a7c092c5@mail.gmail.com>
	<87f2f3b91003111256o451fe7md21aec93c802c0f9@mail.gmail.com>
Message-ID: <FFD74D15-D9F7-408C-AF7A-77C148B0D983@gmail.com>

Thanks All.
The phone isn't the issue, when I push the * I receive a message from freeswitch about mute-on/mute-off. I tried to make current however I get a compile error with the current version see below, this is compiling on Mac OS-X 10.6.2 older version of freeswitch did compile fine.

Compiling /usr/src/freeswitch-snapshot/src/mod/say/mod_say_en/mod_say_en.c ...
Creating mod_say_en.so...
Undefined symbols:
  "_switch_ivr_say_spell", referenced from:
      _en_say in mod_say_en.o
  "_switch_ivr_say_ip", referenced from:
      _en_say in mod_say_en.o
ld: symbol(s) not found
collect2: ld returned 1 exit status


Thanks,
Larry Vatland
Computer World
3015 W. Wisconsin Ave.
Appleton, WI 54914
(920) 733-9547
lvatland at gmail.com



On Mar 11, 2010, at 2:56 PM, Michael Collins wrote:

> 
> 
> On Thu, Mar 11, 2010 at 12:48 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Larry - also see the replies to your previous post on this subject - which I would paraphrase as "make sure the FS version you are using to test is the latest"
> Or, in other words, go to your FreeSWITCH source directory and type "make current" and then go get a cup of coffee. When it is done restart FreeSWITCH and test again...
> -MC
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From irmatov at gmail.com  Thu Mar 11 23:20:42 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Fri, 12 Mar 2010 12:20:42 +0500
Subject: [Freeswitch-users] RFC3578
Message-ID: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>

Hi,

Is there support for RFC3578 in FreeSWITCH? If not, is there any plans
to implement it?

As far as I understand from quick browsing through this RFC, it is
about mapping of ISUP overlap signalling to SIP protocol. In short,
remote device (softswitch for example) sends short (incomplete) number
in INVITE message to freeswitch, which responds with '484 address
incomplete', then softswitch send another INVITE with more digits.
Freeswitch then can send another 484, and after some time (enough
digits collected), proceed with usual call establishment.

-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From skoost at skoost.com  Thu Mar 11 23:36:31 2010
From: skoost at skoost.com (Thangappan Mohan)
Date: 12 Mar 2010 07:36:31 +0000
Subject: [Freeswitch-users] Thangappan Mohan sent you a little gift
Message-ID: <20100312073631.D5C2D17F741@skoismta01.skoost.com>

Thangappan Mohan is on Skoost and sent you a little gift.

To collect your gift, follow the link below:
http://www.skoost.com/?id=114961367_5284543

P.S. This is a safe and innocent gift that Thangappan Mohan sent from Skoost.

Skoost

From gavin.henry at gmail.com  Fri Mar 12 02:22:02 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Mar 2010 10:22:02 +0000
Subject: [Freeswitch-users] UK ring tone
Message-ID: <13ca621c1003120222k5061879ei98863652c6ebd2db@mail.gmail.com>

Hi,

I've checked the internal and external sip profiles and think this
needs to be set in autoload switch file as per:

http://wiki.freeswitch.org/wiki/Variable_transfer_ringback

I want by default a UK ringtone to be heard when I call a DDI that is
configured on the system. At the moment it sounds like I'm calling the
US.

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From brian at freeswitch.org  Fri Mar 12 04:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Mar 2010 06:56:56 -0600
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
Message-ID: <B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>

It already works use the respond application in the dialplan to respond 484

/b

On Mar 12, 2010, at 1:20 AM, Timur Irmatov wrote:

> Hi,
> 
> Is there support for RFC3578 in FreeSWITCH? If not, is there any plans
> to implement it?
> 
> As far as I understand from quick browsing through this RFC, it is
> about mapping of ISUP overlap signalling to SIP protocol. In short,
> remote device (softswitch for example) sends short (incomplete) number
> in INVITE message to freeswitch, which responds with '484 address
> incomplete', then softswitch send another INVITE with more digits.
> Freeswitch then can send another 484, and after some time (enough
> digits collected), proceed with usual call establishment.
> 
> -- 
> Timur Irmatov, xmpp:irmatov at jabber.ru
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Fri Mar 12 07:55:29 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 12 Mar 2010 07:55:29 -0800
Subject: [Freeswitch-users] UK ring tone
Message-ID: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>

You can set it in the dialplan prior to transferring or bridging the  
incoming call.

Sent from my iPhone

On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi,
>
> I've checked the internal and external sip profiles and think this
> needs to be set in autoload switch file as per:
>
> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
>
> I want by default a UK ringtone to be heard when I call a DDI that is
> configured on the system. At the moment it sounds like I'm calling the
> US.
>
> Thanks.
>
> -- http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From ivdreg at gmail.com  Fri Mar 12 08:44:57 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Fri, 12 Mar 2010 18:44:57 +0200
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>
	<4B951E63.6060200@coppice.org>
	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>
	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>
	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>
	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>
	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>
Message-ID: <b437a0e1003120844m52404de8l12c4d8e3299c54ac@mail.gmail.com>

Hi Brian and everyone in FS list,

Believe or not in some countries FAX is legal document in court and email is
not. As a tech guy I hate faxing but I live in world full with lawyer and
again because I am tech person when faxing is broken I have problems :(

P.S. This is just an opinion and not related to this lovely project ;)

2010/3/9 Brian West <brian at freeswitch.org>

> There is no such thing as security when humans are involved in the relay of
> information.  Thats fact.
>
> HIPAA is a joke. PCI is a joke... they all don't get it.  Humans are
> involved its not secure.
>
> /b
>
> On Mar 8, 2010, at 4:39 PM, Gabriel Kuri wrote:
>
> > While I'd love to see FAXing just die, especially with the amount of
> headache associated with it, the reality is that specific organizations
> still rely on it heavily on a daily basis (ie Health Care and
> Banking/Finance Industry). Plus they can legally get away with FAXing
> personal data directly between machines, since the FAX transmission doesn't
> seem to be directly covered by all their respective privacy regultions (ie
> HIPAA, PCI, etc.), unless of course it's a FAX-to-email service. Plus most
> of these organizations just don't have the internal expertise to setup
> end-to-end encrypted email nor the impetus to deal with it, since FAXing
> works fine in their eyes. I think it's going to be around quite a while
> longer unless they actually stop making the FAX machines and force people to
> use something else.
> >
> > Cheers,
> > Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ivanov.maxim at gmail.com  Fri Mar 12 09:30:34 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 12 Mar 2010 17:30:34 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com> 
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com> 
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
Message-ID: <fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>

2010/2/26 Kristian Kielhofner <kristian.kielhofner at gmail.com>:
> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes

Doesn't help =( After getting first USER_BUSY it still makes call via
second part of dialstring.

originate {failure_causes=USER_BUSY}sofia/gateway/panas111/223|sofia/gateway/panas112/223
&playback(local_stream://moh)

Relevant log part is following:

2010-03-12 20:26:00.490396 [DEBUG] switch_ivr_originate.c:1176
variable string 0 = [failure_causes=USER_BUSY]
2010-03-12 20:26:00.494260 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/223 [35e8382d-e3fd-4190-b625-47f61ae6be16]
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:3020
(sofia/internal/223) State Change CS_NEW -> CS_INIT
freeswitch at internal> 2010-03-12 20:26:00.494260 [DEBUG]
switch_core_session.c:1000 Send signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_INIT
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:83 sofia/internal/223
SOFIA INIT
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:111
(sofia/internal/223) State Change CS_INIT -> CS_ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT going to sleep
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING
2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:130 sofia/internal/223
SOFIA ROUTING
2010-03-12 20:26:00.494260 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING going to sleep
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA
2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA going to sleep
2010-03-12 20:26:00.494260 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.517975 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.550450 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [terminated][486]
2010-03-12 20:26:00.550450 [NOTICE] sofia.c:4149 Hangup
sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
2010-03-12 20:26:00.550450 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/223 [KILL]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_HANGUP
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:318 sofia/internal/223
Overriding SIP cause 486 with 486 from the other leg
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:350 Channel
sofia/internal/223 hanging up, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:46
sofia/internal/223 Standard HANGUP, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP going to sleep
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:53
sofia/internal/223 Standard REPORTING, cause: USER_BUSY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING going to sleep
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1137 Session
44644 (sofia/internal/223) Locked, Waiting on external entities
2010-03-12 20:26:00.550450 [DEBUG] switch_ivr_originate.c:2467
Originate Resulted in Error Cause: 17 [USER_BUSY]
2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1155 Session
44644 (sofia/internal/223) Ended
2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1157 Close
Channel sofia/internal/223 [CS_DESTROY]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/223) Running State Change CS_DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:267 sofia/internal/223
SOFIA DESTROY
2010-03-12 20:26:00.550450 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/223 [c8fdd3e0-d905-4f02-8955-d21d0bddf925]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:60
sofia/internal/223 Standard DESTROY
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY going to sleep
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:3020
(sofia/internal/223) State Change CS_NEW -> CS_INIT
2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_INIT
2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT
2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:83 sofia/internal/223
SOFIA INIT
2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:111
(sofia/internal/223) State Change CS_INIT -> CS_ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.554707 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/223) State INIT going to sleep
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING
2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:130 sofia/internal/223
SOFIA ROUTING
2010-03-12 20:26:00.554707 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/223) State ROUTING going to sleep
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA
2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
(sofia/internal/223) State CONSUME_MEDIA going to sleep
2010-03-12 20:26:00.578400 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [calling][0]
2010-03-12 20:26:00.614451 [DEBUG] sofia.c:3583 Channel
sofia/internal/223 entering state [terminated][486]
2010-03-12 20:26:00.614451 [NOTICE] sofia.c:4149 Hangup
sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
2010-03-12 20:26:00.614451 [DEBUG] switch_channel.c:1896 Send signal
sofia/internal/223 [KILL]
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_HANGUP
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP
2010-03-12 20:26:00.614451 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:451
thread mismatch skipping state handler.
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:318 sofia/internal/223
Overriding SIP cause 486 with 486 from the other leg
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:350 Channel
sofia/internal/223 hanging up, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:46
sofia/internal/223 Standard HANGUP, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:478
(sofia/internal/223) State HANGUP going to sleep
2010-03-12 20:26:00.618419 [DEBUG] switch_ivr_originate.c:2467
Originate Resulted in Error Cause: 17 [USER_BUSY]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/223) Running State Change CS_REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:53
sofia/internal/223 Standard REPORTING, cause: USER_BUSY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
(sofia/internal/223) State REPORTING going to sleep
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:319
(sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
signal sofia/internal/223 [BREAK]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1137 Session
44645 (sofia/internal/223) Locked, Waiting on external entities
2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1155 Session
44645 (sofia/internal/223) Ended
2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1157 Close
Channel sofia/internal/223 [CS_DESTROY]
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:415
(sofia/internal/223) Running State Change CS_DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY
2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:267 sofia/internal/223
SOFIA DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:60
sofia/internal/223 Standard DESTROY
2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
(sofia/internal/223) State DESTROY going to sleep



From anthony.minessale at gmail.com  Fri Mar 12 09:40:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 11:40:09 -0600
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com>
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com>
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com>
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com>
	<fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com>
Message-ID: <191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>

you need fail_on_single_reject=true inside the {} as well


On Fri, Mar 12, 2010 at 11:30 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> 2010/2/26 Kristian Kielhofner <kristian.kielhofner at gmail.com>:
> > http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>
> Doesn't help =( After getting first USER_BUSY it still makes call via
> second part of dialstring.
>
> originate
> {failure_causes=USER_BUSY}sofia/gateway/panas111/223|sofia/gateway/panas112/223
> &playback(local_stream://moh)
>
> Relevant log part is following:
>
> 2010-03-12 20:26:00.490396 [DEBUG] switch_ivr_originate.c:1176
> variable string 0 = [failure_causes=USER_BUSY]
> 2010-03-12 20:26:00.494260 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/223 [35e8382d-e3fd-4190-b625-47f61ae6be16]
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:3020
> (sofia/internal/223) State Change CS_NEW -> CS_INIT
> freeswitch at internal> 2010-03-12 20:26:00.494260 [DEBUG]
> switch_core_session.c:1000 Send signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_INIT
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:83 sofia/internal/223
> SOFIA INIT
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:111
> (sofia/internal/223) State Change CS_INIT -> CS_ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] mod_sofia.c:130 sofia/internal/223
> SOFIA ROUTING
> 2010-03-12 20:26:00.494260 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA
> 2010-03-12 20:26:00.494260 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA going to sleep
> 2010-03-12 20:26:00.494260 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.517975 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.550450 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [terminated][486]
> 2010-03-12 20:26:00.550450 [NOTICE] sofia.c:4149 Hangup
> sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_channel.c:1896 Send signal
> sofia/internal/223 [KILL]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_HANGUP
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:318 sofia/internal/223
> Overriding SIP cause 486 with 486 from the other leg
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:350 Channel
> sofia/internal/223 hanging up, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/223 Standard HANGUP, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/223 Standard REPORTING, cause: USER_BUSY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:319
> (sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1137 Session
> 44644 (sofia/internal/223) Locked, Waiting on external entities
> 2010-03-12 20:26:00.550450 [DEBUG] switch_ivr_originate.c:2467
> Originate Resulted in Error Cause: 17 [USER_BUSY]
> 2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1155 Session
> 44644 (sofia/internal/223) Ended
> 2010-03-12 20:26:00.550450 [NOTICE] switch_core_session.c:1157 Close
> Channel sofia/internal/223 [CS_DESTROY]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/223) Running State Change CS_DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:267 sofia/internal/223
> SOFIA DESTROY
> 2010-03-12 20:26:00.550450 [NOTICE] switch_channel.c:613 New Channel
> sofia/internal/223 [c8fdd3e0-d905-4f02-8955-d21d0bddf925]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/223 Standard DESTROY
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY going to sleep
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:3020
> (sofia/internal/223) State Change CS_NEW -> CS_INIT
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_INIT
> 2010-03-12 20:26:00.550450 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT
> 2010-03-12 20:26:00.550450 [DEBUG] mod_sofia.c:83 sofia/internal/223
> SOFIA INIT
> 2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:111
> (sofia/internal/223) State Change CS_INIT -> CS_ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.554707 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/223) State INIT going to sleep
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] mod_sofia.c:130 sofia/internal/223
> SOFIA ROUTING
> 2010-03-12 20:26:00.554707 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/223) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:333
> (sofia/internal/223) State ROUTING going to sleep
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA
> 2010-03-12 20:26:00.554707 [DEBUG] switch_core_state_machine.c:352
> (sofia/internal/223) State CONSUME_MEDIA going to sleep
> 2010-03-12 20:26:00.578400 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [calling][0]
> 2010-03-12 20:26:00.614451 [DEBUG] sofia.c:3583 Channel
> sofia/internal/223 entering state [terminated][486]
> 2010-03-12 20:26:00.614451 [NOTICE] sofia.c:4149 Hangup
> sofia/internal/223 [CS_CONSUME_MEDIA] [USER_BUSY]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_channel.c:1896 Send signal
> sofia/internal/223 [KILL]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_HANGUP
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.614451 [DEBUG] switch_core_state_machine.c:451
> thread mismatch skipping state handler.
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:318 sofia/internal/223
> Overriding SIP cause 486 with 486 from the other leg
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:350 Channel
> sofia/internal/223 hanging up, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/223 Standard HANGUP, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:478
> (sofia/internal/223) State HANGUP going to sleep
> 2010-03-12 20:26:00.618419 [DEBUG] switch_ivr_originate.c:2467
> Originate Resulted in Error Cause: 17 [USER_BUSY]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/223) State Change CS_HANGUP -> CS_REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:306
> (sofia/internal/223) Running State Change CS_REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/223 Standard REPORTING, cause: USER_BUSY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:569
> (sofia/internal/223) State REPORTING going to sleep
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:319
> (sofia/internal/223) State Change CS_REPORTING -> CS_DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1000 Send
> signal sofia/internal/223 [BREAK]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_session.c:1137 Session
> 44645 (sofia/internal/223) Locked, Waiting on external entities
> 2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1155 Session
> 44645 (sofia/internal/223) Ended
> 2010-03-12 20:26:00.618419 [NOTICE] switch_core_session.c:1157 Close
> Channel sofia/internal/223 [CS_DESTROY]
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:415
> (sofia/internal/223) Running State Change CS_DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] mod_sofia.c:267 sofia/internal/223
> SOFIA DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/223 Standard DESTROY
> 2010-03-12 20:26:00.618419 [DEBUG] switch_core_state_machine.c:426
> (sofia/internal/223) State DESTROY going to sleep
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

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pstn:+19193869900
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From ivanov.maxim at gmail.com  Fri Mar 12 10:46:06 2010
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Fri, 12 Mar 2010 18:46:06 +0000
Subject: [Freeswitch-users] Multiple gateways dial string and user busy
In-Reply-To: <191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>
References: <fbfa7b0c1002240813x58ba8541h7fef5b94f045cbda@mail.gmail.com> 
	<87f2f3b91002251149m13c0d961k722dd3375f46b6e8@mail.gmail.com> 
	<fbfa7b0c1002260823l11b8d609lf4162eeb61ffc036@mail.gmail.com> 
	<4d15ff861002260838v6aa624a1t398eb50bddc75ab1@mail.gmail.com> 
	<fbfa7b0c1003120930y4b389318peb58d76643e118b5@mail.gmail.com> 
	<191c3a031003120940v65567d54w827a5ff0ec069e59@mail.gmail.com>
Message-ID: <fbfa7b0c1003121046t3091f1d1l1dc6d1815a766eb8@mail.gmail.com>

> you need fail_on_single_reject=true inside the {} as well

brilliant! works like a charm!



From sean at obscuradigital.com  Fri Mar 12 14:18:00 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Fri, 12 Mar 2010 14:18:00 -0800
Subject: [Freeswitch-users] DTMF problem
Message-ID: <C7BFFD98.64D5%sean@obscuradigital.com>

    Question:

I have FS 1.0.5 installed on a Centos 5.4 server box.  I hope there?s a
simple answer to this question.

I?m having problems in particular when connecting to a outside conference
line on a completely different phone system entering numbers that the
receiver should accept.  For example, call into a conference line,
auto-attendent ask for the pin number, I then enter pin number, but The
receiving ivr tells me I?ve entered the wrong pin number.  I?m assuming that
I?ve reached a timeout and the ivr is letting me know that I?ve entered the
incorrect pin number.  My thinking is that this is a DTMF issue.  I?m new to
FS so not sure if I need to enable something in my dialplan or not.

Thoughts?
Thanks in advance
Sean
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From anthony.minessale at gmail.com  Fri Mar 12 14:32:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 16:32:48 -0600
Subject: [Freeswitch-users] DTMF problem
In-Reply-To: <C7BFFD98.64D5%sean@obscuradigital.com>
References: <C7BFFD98.64D5%sean@obscuradigital.com>
Message-ID: <191c3a031003121432l6fa95adcvc1693a268f9b289@mail.gmail.com>

Yet its quite possible.  If you are using a provider it can be a problem
with them as well.
There are numerous methods to debug DTMF issues on our wiki page.


On Fri, Mar 12, 2010 at 4:18 PM, Sean Holt <sean at obscuradigital.com> wrote:

>     Question:
>
> I have FS 1.0.5 installed on a Centos 5.4 server box.  I hope there?s a
> simple answer to this question.
>
> I?m having problems in particular when connecting to a outside conference
> line on a completely different phone system entering numbers that the
> receiver should accept.  For example, call into a conference line,
> auto-attendent ask for the pin number, I then enter pin number, but The
> receiving ivr tells me I?ve entered the wrong pin number.  I?m assuming that
> I?ve reached a timeout and the ivr is letting me know that I?ve entered the
> incorrect pin number.  My thinking is that this is a DTMF issue.  I?m new to
> FS so not sure if I need to enable something in my dialplan or not.
>
> Thoughts?
> Thanks in advance
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From gavin.henry at gmail.com  Fri Mar 12 14:38:17 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 12 Mar 2010 22:38:17 +0000
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
Message-ID: <13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>

Ok, not system wide?

Thanks,

Gavin.

On 12/03/2010, Michael S  Collins <msc at freeswitch.org> wrote:
> You can set it in the dialplan prior to transferring or bridging the
> incoming call.
>
> Sent from my iPhone
>
> On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:
>
>> Hi,
>>
>> I've checked the internal and external sip profiles and think this
>> needs to be set in autoload switch file as per:
>>
>> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
>>
>> I want by default a UK ringtone to be heard when I call a DDI that is
>> configured on the system. At the moment it sounds like I'm calling the
>> US.
>>
>> Thanks.
>>
>> -- http://www.suretecsystems.com/services/openldap/
>> http://www.suretectelecom.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From anthony.minessale at gmail.com  Fri Mar 12 15:05:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Mar 2010 17:05:40 -0600
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
Message-ID: <191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>

you can set it globally in vars.xml too


On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> Ok, not system wide?
>
> Thanks,
>
> Gavin.
>
> On 12/03/2010, Michael S  Collins <msc at freeswitch.org> wrote:
> > You can set it in the dialplan prior to transferring or bridging the
> > incoming call.
> >
> > Sent from my iPhone
> >
> > On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry at gmail.com> wrote:
> >
> >> Hi,
> >>
> >> I've checked the internal and external sip profiles and think this
> >> needs to be set in autoload switch file as per:
> >>
> >> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
> >>
> >> I want by default a UK ringtone to be heard when I call a DDI that is
> >> configured on the system. At the moment it sounds like I'm calling the
> >> US.
> >>
> >> Thanks.
> >>
> >> -- http://www.suretecsystems.com/services/openldap/
> >> http://www.suretectelecom.com
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> Sent from my mobile device
>
> http://www.suretecsystems.com/services/openldap/
> http://www.suretectelecom.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Fri Mar 12 15:09:29 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Mar 2010 17:09:29 -0600
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
	<191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
Message-ID: <FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>

in fact an example for uk-ring is already there.

/b

On Mar 12, 2010, at 5:05 PM, Anthony Minessale wrote:

> you can set it globally in vars.xml too
> 
> 
> On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
> Ok, not system wide?
> 
> Thanks,
> 
> Gavin.

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From m.sobkow at marketelsystems.com  Fri Mar 12 15:48:25 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Fri, 12 Mar 2010 17:48:25 -0600
Subject: [Freeswitch-users] OpenZap/Sangoma drivers
Message-ID: <4B9AD2C9.1010004@marketelsystems.com>

I filed a bug report this morning because of a Freeswitch build problem, 
but the resolution was to update the OpenZap/Sangoma drivers.  Updating 
the Sangoma drivers was a little hairy because of our project build 
process needing to be updated, but I got it figured out and got the 
latest build of Freeswitch installed (svn checkout this morning.)  Next 
I need to re-test the call recordings.



From fs-list at communicatefreely.net  Fri Mar 12 18:38:08 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 12 Mar 2010 21:38:08 -0500
Subject: [Freeswitch-users] voicemail_greeting_number stickiness
Message-ID: <4B9AFA90.6050409@communicatefreely.net>


Hello all,

I'm trying to set up our voice mail system so that greetings can be mapped to failure conditions,
ie. "I'm in my office, but not taking calls" when a phone is busy or in DND mode, but "Sorry I'm not
here" if no_answer.

I programmed things to set voicemail_greeting_number in the dialplan based on the bridge failure
cause set when we rang the phone, but when I execute the voicemail application, it stores that
greeting number in the database.  I would much rather override for this call only, so that the user
can still take advantage of the voicemail application's option to select a greeting.  (which would
be used as the no answer greeting under normal circumstances).

Any thoughts?

-Tim



From jmesquita at freeswitch.org  Fri Mar 12 20:23:19 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 13 Mar 2010 01:23:19 -0300
Subject: [Freeswitch-users] voicemail_greeting_number stickiness
In-Reply-To: <4B9AFA90.6050409@communicatefreely.net>
References: <4B9AFA90.6050409@communicatefreely.net>
Message-ID: <ebbc2d551003122023r22c0ed0bx580dad9893e21a1c@mail.gmail.com>

Use hangup hook to set it back to whatever is default?

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions
t: +1 (646) 4959927


On Fri, Mar 12, 2010 at 11:38 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

>
> Hello all,
>
> I'm trying to set up our voice mail system so that greetings can be mapped
> to failure conditions,
> ie. "I'm in my office, but not taking calls" when a phone is busy or in DND
> mode, but "Sorry I'm not
> here" if no_answer.
>
> I programmed things to set voicemail_greeting_number in the dialplan based
> on the bridge failure
> cause set when we rang the phone, but when I execute the voicemail
> application, it stores that
> greeting number in the database.  I would much rather override for this
> call only, so that the user
> can still take advantage of the voicemail application's option to select a
> greeting.  (which would
> be used as the no answer greeting under normal circumstances).
>
> Any thoughts?
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fs-list at communicatefreely.net  Fri Mar 12 21:34:50 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sat, 13 Mar 2010 00:34:50 -0500
Subject: [Freeswitch-users] Gateway context for incoming calls
Message-ID: <4B9B23FA.6080103@communicatefreely.net>

Hello,

Is there a way for incoming calls from a gateway to go to a context other than the default one set
in the sofia profile?  I need a way to separate calls from providers vs. calls from the public at large.

Here's the scenario:

Our external sofia profile maps to the public IP address and listens on port 5060.  Our providers
send calls to this address, but we also need to have anonymous sip calls come in to the public
address, on the default port.  We want the calls from our SIP trunks to get billed appropriately,
and increment our mod_limit counters, etc.  If a call comes in from somewhere other than a SIP
trunk, we want it to have access to extensions, but be rated as a free call.

All our SIP trunks are mapped using static addresses - there isn't any registration or
authentication happening in either direction.  Each one has a gateway entry.

Thanks!

-Tim



From msc at freeswitch.org  Fri Mar 12 22:59:20 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 12 Mar 2010 22:59:20 -0800
Subject: [Freeswitch-users] Gateway context for incoming calls
In-Reply-To: <4B9B23FA.6080103@communicatefreely.net>
References: <4B9B23FA.6080103@communicatefreely.net>
Message-ID: <87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>

Try sending one of these calls to the info application in your public
dialplan. Most likely there will be a few fields which you can key off of in
your dialplan conditions. If your providers have static IPs then you can
always use the network_addr field to match on. Once you match on a
particular IP address (or whichever field) then you can execute the transfer
app and send the call to another context altogether:

<extension name="example IP addr match">
    <condition field="network_addr" expression="^1\.2\.3\.4$"/> <!-- IP addr
of originating server -->
    <condition field="destination_number" expression="^(.*)$">
         <action application="transfer" data="$1 XML my_context"/>
    </condition>
</extension>

Let us know how it goes.
-MC

On Fri, Mar 12, 2010 at 9:34 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> Is there a way for incoming calls from a gateway to go to a context other
> than the default one set
> in the sofia profile?  I need a way to separate calls from providers vs.
> calls from the public at large.
>
> Here's the scenario:
>
> Our external sofia profile maps to the public IP address and listens on
> port 5060.  Our providers
> send calls to this address, but we also need to have anonymous sip calls
> come in to the public
> address, on the default port.  We want the calls from our SIP trunks to get
> billed appropriately,
> and increment our mod_limit counters, etc.  If a call comes in from
> somewhere other than a SIP
> trunk, we want it to have access to extensions, but be rated as a free
> call.
>
> All our SIP trunks are mapped using static addresses - there isn't any
> registration or
> authentication happening in either direction.  Each one has a gateway
> entry.
>
> Thanks!
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Sat Mar 13 00:51:13 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 13 Mar 2010 19:51:13 +1100
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
Message-ID: <20100313085113.GA32174@jdc.jasonjgw.net>

Code has now been posted at https://developer.skype.com/silk/

It appears from that page that there may be patent issues for anything other
than testing of the codec, although it is stated elsewhere that the licence
terms are royalty-free.

To build the module (for those planning to do the aforementioned testing) you
need to run make in libs/silk first to compile the library, then build
FreeSWITCH; this will change once the build system is updated appropriately.




From steveu at coppice.org  Sat Mar 13 02:16:51 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 13 Mar 2010 18:16:51 +0800
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <20100313085113.GA32174@jdc.jasonjgw.net>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>	<20100310051344.GA19653@jdc.jasonjgw.net>	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
	<20100313085113.GA32174@jdc.jasonjgw.net>
Message-ID: <4B9B6613.4040901@coppice.org>

On 03/13/2010 04:51 PM, Jason White wrote:
> Code has now been posted at https://developer.skype.com/silk/
>
> It appears from that page that there may be patent issues for anything other
> than testing of the codec, although it is stated elsewhere that the licence
> terms are royalty-free.
>    
That sounds like they are trying for a similar arrangement Polycom has 
with G.722.1. You can use G.722.1 royalty free, if you ask for a licence 
first and comply with some simple conditions - mostly ensuring that you 
properly follow the G.722.1 spec. It works well enough for most 
purposes, though it conflicts with some free licences, and there have 
been some disagreements. For example, Polycom said it was OK for 
Freeswitch to be distributed with the source code for G.722.1 bundled. 
It seems Digium believe Asterisk may not be distributed with the source 
code for G.722.1.
> To build the module (for those planning to do the aforementioned testing) you
> need to run make in libs/silk first to compile the library, then build
> FreeSWITCH; this will change once the build system is updated appropriately.
>
>    
Steve




From brian at freeswitch.org  Sat Mar 13 08:08:37 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 10:08:37 -0600
Subject: [Freeswitch-users] Gateway context for incoming calls
In-Reply-To: <87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>
References: <4B9B23FA.6080103@communicatefreely.net>
	<87f2f3b91003122259pabf3d67t48550db89ea0c808@mail.gmail.com>
Message-ID: <47ED6B59-7300-467B-B879-643266018395@freeswitch.org>

Its worse.. the only way you can map an inbound call back to the gateway in question is the far side has to call the registered contact you sent on outbound registration.  Most services don't follow this rule.

/b

On Mar 13, 2010, at 12:59 AM, Michael Collins wrote:

> Try sending one of these calls to the info application in your public dialplan. Most likely there will be a few fields which you can key off of in your dialplan conditions. If your providers have static IPs then you can always use the network_addr field to match on. Once you match on a particular IP address (or whichever field) then you can execute the transfer app and send the call to another context altogether:
> 
> <extension name="example IP addr match">
>     <condition field="network_addr" expression="^1\.2\.3\.4$"/> <!-- IP addr of originating server -->
>     <condition field="destination_number" expression="^(.*)$">
>          <action application="transfer" data="$1 XML my_context"/>
>     </condition>
> </extension>
> 
> Let us know how it goes.
> -MC




From gavin.henry at gmail.com  Sat Mar 13 11:20:31 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 13 Mar 2010 19:20:31 +0000
Subject: [Freeswitch-users] UK ring tone
In-Reply-To: <FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>
References: <B99FC8DB-579D-4CD0-882F-7E92761A434C@freeswitch.org>
	<13ca621c1003121438n5974d599sdaac4bb40211de74@mail.gmail.com>
	<191c3a031003121505q346a363dv87103eda6cbf6f1b@mail.gmail.com>
	<FAAC81EE-2F87-460A-B7B7-F84ADB879E23@freeswitch.org>
Message-ID: <13ca621c1003131120n4a322ad1mfa7b910f593b7b5c@mail.gmail.com>

Ok, thanks. I think there is a us and fr one by default. Will comment
out those and test.

Thanks,

Gavin.

On 12/03/2010, Brian West <brian at freeswitch.org> wrote:
> in fact an example for uk-ring is already there.
>
> /b
>
> On Mar 12, 2010, at 5:05 PM, Anthony Minessale wrote:
>
>> you can set it globally in vars.xml too
>>
>>
>> On Fri, Mar 12, 2010 at 4:38 PM, Gavin Henry <gavin.henry at gmail.com>
>> wrote:
>> Ok, not system wide?
>>
>> Thanks,
>>
>> Gavin.
>
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com



From jbrucehopkins at gmail.com  Sat Mar 13 14:56:50 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sat, 13 Mar 2010 22:56:50 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <000001cabfbb$f2114c90$d633e5b0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
Message-ID: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>

Hi Roly,

I promised I'd let you know when I got it working - so here it is:

As Brian said, doublenat is no longer needed.

It looks like I must have just been confusing things horribly by using
various things I'd read about, including:

 <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

<param name="aggressive-nat-detection" value="true"/>


These, and who knows what other mucking about I had been doing, must have
confused everything as none of this is now necessary.


All I did - after a reinstall - to get everything working through double
NATs is:


1.  In vars.xml

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>


 <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>


2.  Then in prefix/sip_profiles/internal.xml

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

 <param name="ext-sip-ip" value="$${external_sip_ip)"/>


Then as the man says, it just works.

I decided not to enable options pings to keep the far-end NAT open to enable
calling the remote phone - instead I configured it to send keep-alives.


And that's it.  I can't believe the knots I was tying myself in.  It just
works !

cheers

Bruce








On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Mar 13 17:05:40 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 19:05:40 -0600
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
Message-ID: <3B76DFA1-6304-4B84-A478-387C1CED9D3C@freeswitch.org>

Its a common thing to try too hard.  I personally put in countless hours to make this stuff JUST work. ;)  Also if you're traversing the nat and you expect to talk to clients behind the nat and outside the nat on the same profile you will be required to prefix both ext-rtp-ip and ext-sip-ip with "autonat:<my.fs.ext.ip>" and make sure local-network-acl is set to localnet.auto... I think thats the one in the defaults too.  This gives FreeSWITCH the ability to dynamically lie to the client about which IP is which so that the same profile can talk to clients on either side of the nat.

/b

On Mar 13, 2010, at 4:56 PM, Bruce Hopkins wrote:

> And that's it.  I can't believe the knots I was tying myself in.  It just works !




From mattdfong at gmail.com  Sat Mar 13 17:24:48 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 13 Mar 2010 17:24:48 -0800
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback 	doesn't seem to get called
In-Reply-To: <5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
	<5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
Message-ID: <4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>

Hi Michael,

the cell phone carriers in the US use a beep tone that is out of the range
that is typically detected. i've had success using tone_detect rather than
mod_vmd. if my memory serves me correctly 786 is the tone you need to listen
for. the same is true for sprint and t-mobile voice mail beeps

--matt
freeswitch based voice broadcasting -
http://www.hellohunter.com/voice_broadcast.php
freeswitch based predictive dialing -
http://www.hellohunter.com/predictive_dialer.php

On Wed, Mar 3, 2010 at 7:04 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Has anyone experienced issues with vmd or dtmf with Verizon wireless?  I
> have another script that doesn't even register digits pressed -- it's like
> no keys are pressed at all.
>
> I've checked the FS log and it doesn't seem to log anything with regard to
> the mod_vmd other than an indication of MIN_TIME of 8000 (not sure what that
> does exactly).
>
> Does anyone have any sample scripts available other than what's on the FS
> wiki?
>
>
> On Mon, Mar 1, 2010 at 10:17 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> I've got the following Lua script working in a sense, but the
>> InputCallback never seems to get called while the file is being streamed to
>> the call recipient.  I've tried moving the "vmd start" command and set input
>> callback around a bit, but to no avail.  I'm testing this against a cell
>> phone voice mailbox (Verizon).
>>
>>
>>> freeswitch.consoleLog("info","########################################################\n\n");
>>>
>>>  number_to_call = argv[1]
>>> message_to_play = "/opt/freeswitch/recordings/messages/" .. argv[2]
>>>
>>> voicemail_detected = false;
>>>
>>> function onInput(s, type, obj)
>>>     freeswitch.consoleLog("notice","*********** Type?: " .. type .. "
>>> *************\n");
>>>     -- freeswitch.consoleLog("notice","*********** VMD?: " ..
>>> session:getVariable("vmd_detect") .. " *************\n");
>>>
>>>      if(type == "event" and voicemail_detected == false) then
>>>         freeswitch.consoleLog("notice","************ VOICE MAIL/ANSWERING
>>> MACHINE DETECTED *************\n");
>>>         voicemail_detected = true;
>>>         return "break";
>>>     end
>>> end
>>>
>>> function playbackMessage()
>>>     sleep_time = 1000;
>>>     if(voicemail_detected) then
>>>        sleep_time = 2500;
>>>     end
>>>     -- sleep a second
>>>     session:sleep(sleep_time);
>>>     -- play a file
>>>     session:streamFile(message_to_play);
>>> end
>>>
>>> function notify()
>>>     session =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/voicenetwork/1"
>>> .. number_to_call)
>>>
>>>
>>>
>>>     if(session:ready()) then
>>>         -- answer the call
>>>         session:answer();
>>>         session:setInputCallback("onInput", "");
>>>         session:execute("vmd","start");
>>>
>>>         playbackMessage();
>>>         if(voicemail_detected) then
>>>             freeswitch.consoleLog("notice","************ DOING PLAYBACK
>>> FOR VOICEMAIL/ANSWERING MACHINE *************\n");
>>>             playbackMessage();
>>>         end
>>>
>>>         freeswitch.consoleLog("notice", "********* hanging up session
>>> **********\n");
>>>         -- hangup
>>>         session:hangup();
>>>     end
>>> end
>>>
>>> notify();
>>>
>>>
>>> freeswitch.consoleLog("info","########################################################\n\n");
>>>
>>
>> --
>> Michael De Lorenzo
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Sat Mar 13 17:53:20 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 12:53:20 +1100
Subject: [Freeswitch-users] not answering calls on TDM400 port
Message-ID: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>

Hello,

I noticed Freeswitch wasn't answering calls on one of the TDM400 ports
yesterday.

All other ports seemed to be OK.

Nothing showed up on the console, set at debug log level.

I restarted Freeswitch and it came good.

What should I do if it happens again? Is there any debugging
information I can extract from the system? If so how?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Sat Mar 13 18:47:21 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Mar 2010 20:47:21 -0600
Subject: [Freeswitch-users] Lua Script with mod_vmd,
	setInputCallback  doesn't seem to get called
In-Reply-To: <4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>
References: <5c9dcbfb1003011917h393ddde5ha8cc3ec74d0a4d53@mail.gmail.com>
	<5c9dcbfb1003031904t4586f160ge0daf8c6d07331a7@mail.gmail.com>
	<4256bf831003131724o324ac8e4m7d6b8a8af70e81c@mail.gmail.com>
Message-ID: <4EC6125F-683B-409D-B7D6-C8842586BAD2@freeswitch.org>

You'll need to file a jira.  Include a script that exhibits the behavior.  You'll also need to document it fully and the dialplan entry you're using to produce this issue.

/b

On Mar 13, 2010, at 7:24 PM, Matthew Fong wrote:

> Hi Michael,
> 
> the cell phone carriers in the US use a beep tone that is out of the range that is typically detected. i've had success using tone_detect rather than mod_vmd. if my memory serves me correctly 786 is the tone you need to listen for. the same is true for sprint and t-mobile voice mail beeps
> 
> --matt
> freeswitch based voice broadcasting - http://www.hellohunter.com/voice_broadcast.php
> freeswitch based predictive dialing - http://www.hellohunter.com/predictive_dialer.php

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From brian at microcomaustralia.com.au  Sat Mar 13 20:02:25 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:02:25 +1100
Subject: [Freeswitch-users] Possible typo on Openzap.conf wiki page
	specifying fxs/fxo-channels
In-Reply-To: <cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>
References: <2C33D9D6C1444AAC80950CDF8D38A428@greyhawk.tonecommander.com>
	<3c5cf5261002231524n4e9af307y80b3accb1ee4ff26@mail.gmail.com>
	<a689f57fe216d671d7d1ff132c6b636e@thom.fr.eu.org>
	<3c5cf5261002271711t59294b4fm9aebfdf5b48a6b31@mail.gmail.com>
	<cd27dd8cd4d3869b3b3b341581808673@thom.fr.eu.org>
Message-ID: <3c5cf5261003132002v715c25a3o945baf6062f4f20@mail.gmail.com>

On 1 March 2010 19:10, Fran?ois Legal <devel at thom.fr.eu.org> wrote:
> On this one, I would have to check in the code. I don't know if the span
> order here makes a difference (my guess is no).

Difference isn't so much the order, but one block has tonegroup set (I
assume this is for an extension), and the other doesn't.
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sat Mar 13 20:11:07 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:11:07 +1100
Subject: [Freeswitch-users] internal/external profiles
In-Reply-To: <87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>
References: <3c5cf5261002231627s1dfb04b2m64bf9bfeb3ff171d@mail.gmail.com>
	<87f2f3b91002240818q9f95269lc2cee35e8ac60498@mail.gmail.com>
Message-ID: <3c5cf5261003132011l627f91a8wd953aeb428493fca@mail.gmail.com>

On 25 February 2010 03:18, Michael Collins <msc at freeswitch.org> wrote:
> No problem. Separating them has a few advantages: security, scalability, and
> readability. The first one on the list is definitely the most important. If
> you stuff everything in the internal profile it's easier to open yourself up
> to toll fraud.

Hello,

I am still confused as to how using the external profile can help
reduce toll fraud.

On my system both internal and external have:

<param name="context" value="public"/>

Also, what happens when freeswitch receives an anonymous connection?
Are anonymous connections allowed or rejected? Where is that behaviour
controlled?

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at microcomaustralia.com.au  Sat Mar 13 20:15:00 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 14 Mar 2010 15:15:00 +1100
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <4B8A300C.4060805@xpirio.com>
References: <4B857226.10308@xpirio.com>
	<191c3a031002241058y517e63fdte0d929b97a33f9ac@mail.gmail.com>
	<4B8A300C.4060805@xpirio.com>
Message-ID: <3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>

2010/2/28 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> problem solved with -vm
> with this option we now have the usual low load for 50-70 conference users

I can't see any documentation for vm, either here:

http://wiki.freeswitch.org/wiki/Command_line

or here:

huey:/opt/freeswitch/conf# /opt/freeswitch/bin/freeswitch --help
these are the optional arguments you can pass to freeswitch
	-nf                    -- no forking
	-u [user]              -- specify user to switch to
	-g [group]             -- specify group to switch to
	-help                  -- this message
	-version               -- print the version and exit
	-waste                 -- allow memory waste
	-core                  -- dump cores
	-hp                    -- enable high priority settings
	-vg                    -- run under valgrind
	-nosql                 -- disable internal sql scoreboard
	-heavy-timer           -- Heavy Timer, possibly more accurate but at a cost
	-nonat                 -- disable auto nat detection
	-nocal                 -- disable clock calibration
	-nort                  -- disable clock clock_realtime
	-stop                  -- stop freeswitch
	-nc                    -- do not output to a console and background
	-c                     -- output to a console and stay in the foreground
	-conf [confdir]        -- specify an alternate config dir
	-log [logdir]          -- specify an alternate log dir
	-run [rundir]          -- specify an alternate run dir
	-db [dbdir]            -- specify an alternate db dir
	-mod [moddir]          -- specify an alternate mod dir
	-htdocs [htdocsdir]    -- specify an alternate htdocs dir
	-scripts [scriptsdir]  -- specify an alternate scripts dir



What does it do?
-- 
Brian May <brian at microcomaustralia.com.au>



From sos at sokhapkin.dyndns.org  Sat Mar 13 20:26:28 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 13 Mar 2010 23:26:28 -0500
Subject: [Freeswitch-users] conferences lead to high server load
In-Reply-To: <3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>
References: <4B857226.10308@xpirio.com> <4B8A300C.4060805@xpirio.com>
	<3c5cf5261003132015q65c61f6cq5a24b5540912c92d@mail.gmail.com>
Message-ID: <201003132326.28390.sos@sokhapkin.dyndns.org>

-vm option has been removed and is the default behavior now, -heavy-timer 
option switches to old (pre-1.0.5) way.

On Saturday 13 March 2010, Brian May wrote:
> 2010/2/28 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> > problem solved with -vm
> > with this option we now have the usual low load for 50-70 conference
> > users
> 
> I can't see any documentation for vm, either here:
> 
> http://wiki.freeswitch.org/wiki/Command_line
> 
> or here:
> 
> huey:/opt/freeswitch/conf# /opt/freeswitch/bin/freeswitch --help
> these are the optional arguments you can pass to freeswitch
> 	-nf                    -- no forking
> 	-u [user]              -- specify user to switch to
> 	-g [group]             -- specify group to switch to
> 	-help                  -- this message
> 	-version               -- print the version and exit
> 	-waste                 -- allow memory waste
> 	-core                  -- dump cores
> 	-hp                    -- enable high priority settings
> 	-vg                    -- run under valgrind
> 	-nosql                 -- disable internal sql scoreboard
> 	-heavy-timer           -- Heavy Timer, possibly more accurate but at a
>  cost -nonat                 -- disable auto nat detection
> 	-nocal                 -- disable clock calibration
> 	-nort                  -- disable clock clock_realtime
> 	-stop                  -- stop freeswitch
> 	-nc                    -- do not output to a console and background
> 	-c                     -- output to a console and stay in the foreground
> 	-conf [confdir]        -- specify an alternate config dir
> 	-log [logdir]          -- specify an alternate log dir
> 	-run [rundir]          -- specify an alternate run dir
> 	-db [dbdir]            -- specify an alternate db dir
> 	-mod [moddir]          -- specify an alternate mod dir
> 	-htdocs [htdocsdir]    -- specify an alternate htdocs dir
> 	-scripts [scriptsdir]  -- specify an alternate scripts dir
> 
> 
> 
> What does it do?
> 




From nepaligas at yahoo.com  Sat Mar 13 22:22:14 2010
From: nepaligas at yahoo.com (Prabin Shrestha)
Date: Sat, 13 Mar 2010 22:22:14 -0800 (PST)
Subject: [Freeswitch-users] problem in sending calls to internal Gateway.
Message-ID: <249845.71206.qm@web65402.mail.ac4.yahoo.com>

hello,

My extension are working properly. But they can't send calls to gateway. I have 2 interfaces, one with public and another with private lan. Gateway is in private lan. Please suggest.
My configurations are as follows:

/conf/sip_profiles/external.xml:
    <param name="rtp-ip" value="110.4.12.111"/>
    <param name="sip-ip" value="110.4.12.111"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

/conf/sip_profiles/internal.xml:
     <param name="rtp-ip" value="192.168.1.11"/>
     <param name="sip-ip" value="192.168.1.11"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="apply-nat-acl" value="nat.auto"/>

haven't made any changes in conf/autoload_configs/acl.conf.xml

to route calls from providers, /dialplan/public/providers.xml:
    <extension name="provider">
    <condition field="network_addr" expression="200.144.44.177"/>
    <condition field="destination_number" expression="^(\d+)$">
        <action application="transfer" data="$1 XML context_AX"/>
    </condition>
   </extension>

gateway is defined at /conf/sip_profiles/internal/gateways.xml:
   <include>
   <gateway name="gateway_AX">
    <param name="username" value="admin"/>
    <param name="password" value="nepal123"/>
    <param name="proxy" value="192.168.1.100"/>
    <param name="register" value="false"/>
    <param name="extension" value="1001"/>
    </gateway>
   </include>


To route internal calls from extensions, /conf/dialplan/default.xml:
<extension name="my_internal">
    <!--<condition field="${ruleset}" expression="internal" /> -->
    <condition field="destination_number" expression="^(d+)$">
      <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
    </condition>
  </extension>

to route calls from providers, conf/dialplan/inbound_routing.xml:
<context name="context_AX">
    <extension name="from_provider_to_AX">    
    <!--<condition field="destination_number" expression="^00(1.*)$">-->
    <condition field="destination_number" expression="^9(\d+)$">    
    <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
    </condition>
    </extension>
</context>

Please help.


      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/




From dome at tel.co.th  Sun Mar 14 05:37:04 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 14 Mar 2010 19:37:04 +0700
Subject: [Freeswitch-users] Inline and cond
Message-ID: <8ccbff061003140537r550234d2j6c6124a6916ea8ec@mail.gmail.com>

I try to use cond in my dialplan

     <condition field="destination_number" expression="^(1111)">
        <action application="answer"/>
        <action inline="true" application="set" data="b=25"/>
     </condition>
     <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$">
        <action application="log" data="ALERT ${b} is indeed greater than 12"/>
        <anti-action application="log" data="ALERT ${b} is not greater
than 12"/>
     </condition>

it's work but when use set b=${b1} it's not work

     <condition field="destination_number" expression="^(1111)">
        <action application="answer"/>
        <action inline="true" application="set" data="b1=25"/>
        <action inline="true" application="set" data="b=${b1}"/>
     </condition>
     <condition field="${cond(${b} > 12 ? YES : NO)}" expression="^YES$">
        <action application="log" data="ALERT ${b} is indeed greater than 12"/>
        <anti-action application="log" data="ALERT ${b} is not greater
than 12"/>
     </condition>

What's wrong ? inline or cond ?

Dome C.



From yehavi.bourvine at gmail.com  Sun Mar 14 08:18:23 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 14 Mar 2010 17:18:23 +0200
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a call
	is left on hold?
Message-ID: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>

Hello,

   We have a strange behavior which my users told me today, but I have a
feeling it lasts for a long time (we are running now version 16972).

User A calls user B on a Polycom. User B press the "transfer" key and calls
C who answers. B then press "transfer" again, and now A talks to C. However,
B is left with a held call to the voicemail of C...

Anyone saw this behaviour?  unattended transfer and conference work ok.

                             Thanks! __Yehavi:
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From brian at freeswitch.org  Sun Mar 14 10:50:28 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:50:28 -0500
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
Message-ID: <C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>

Are you doing this with the latest SVN and default dialplan or are you doing something out of the usual?

/b

On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:

> Hello,
>  
>    We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972).
>  
> User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C...
>  
> Anyone saw this behaviour?  unattended transfer and conference work ok.
>  
>                              Thanks! __Yehavi: 




From brian at freeswitch.org  Sun Mar 14 10:53:55 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:53:55 -0500
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
Message-ID: <86173DCE-6BE0-4663-B90E-81244587F4C4@freeswitch.org>

I can confirm I get the proper behavior.

/b

On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:

> Hello,
>  
>    We have a strange behavior which my users told me today, but I have a feeling it lasts for a long time (we are running now version 16972).
>  
> User A calls user B on a Polycom. User B press the "transfer" key and calls C who answers. B then press "transfer" again, and now A talks to C. However, B is left with a held call to the voicemail of C...
>  
> Anyone saw this behaviour?  unattended transfer and conference work ok.
>  
>                              Thanks! __Yehavi: 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From quentezcorp at gmail.com  Fri Mar 12 13:03:28 2010
From: quentezcorp at gmail.com (Quentin CALVEZ)
Date: Fri, 12 Mar 2010 22:03:28 +0100
Subject: [Freeswitch-users] js.dll crash on Windows Server 2008 R2 x64
Message-ID: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>

Hello everybody,

I've been trying Freeswitch those last days and there is one error that I
don't know what to do about. I'm using FusionPBX to manage my freeswitch
installation and it works great, however, whenever I try to use a function
involving javascript (auto atendants, hunt groups...) the server crashs
telling me that the faulty module is js32.dll.

I really don't know what to do about it. Could someone help me :) ?

Regards.

Quentin Calvez
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From roland at haenel.me  Sat Mar 13 12:38:26 2010
From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=)
Date: Sat, 13 Mar 2010 21:38:26 +0100
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
Message-ID: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>

 Hello,

I'm developing a very simple call-through application using FreeSWITCH, but
currently
I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
header
for outgoing calls if I want so signal an arbitrary caller ID in the 'From'
field.

Setup is as follows:
   - a-leg comes in from an external gateway
   - dialplan dispatches the call to a perl handler script
   - perl script looks like this (simplified as much as possible for this
description):

            $num = "02222222";
            $session->answer();
            $session->setVariable("effective_caller_id_name", "The
Redirector");
            $session->setVariable("sip_h_P-Asserted-Identity", '<
sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
            $session->execute("bridge", "sofia/gateway/mygateway/$num);

The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity
header as
set in the perl script. But there is already an existing P-Asserted-Identity
header by default,
so I end up with two of them. This is the outgoing INVITE (b-leg):


INVITE sip:02222222 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
Max-Forwards: 63
From: "The Redirector" <sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
>;tag=DBSvZ9NZc750g
To: <sip:02222222 at x.x.x.x>
Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
CSeq: 128071597 INVITE
Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 244
X-Port: 5060
X-FS-Support: update_display
P-Asserted-Identity: "The Redirector"
<sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>
P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
>

[... sdp ...]

Numbers are:

    01111111       - the original caller (a-leg & b-leg From:)
    02222222       - the number redirected to (b-leg To:)
    0123456789    - the number my provider needs in P-Asserted-Identity
('redirector's number')

So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
line, but unfortunately it already includes another P-Asserted-Identity
011111111
line, and that breaks the setup at my provider.

I really appreciate any help.

Greetings,
Roland

--
Roland Haenel
QSC AG - http://www.qsc.de
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From rentmycoder at gmail.com  Sun Mar 14 10:40:02 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Sun, 14 Mar 2010 18:40:02 +0100
Subject: [Freeswitch-users] how to access session variables after hangup???
Message-ID: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>

Hi guys,
Freeswitch rocks!
except the documentation:(
I'm not able to figure out how to access session variables after hangup...
I try to dial out on a gateway and detect it the called party hangs up
before bridge...

bgapi originate
{ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
hangup.lua ${uuid}'}sofia/gateway/phoneno &park()

I cannot access the variables using:
1. script arguments:
'luarun hangup.lua ${uuid}'}
2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
(${api_hangup_hook} contains a variable)

2. session:getVariable("originate_disposition")
2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
to call global 'getVar' (a nil value)

3. params:getHeader("variable_sip_req_uri")
2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
to index global 'params' (a nil value)

I have asked this before on the dev list too, but Anthony didn't gave
me a clean answer here:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html

"In the script you can access the data from the env event which is
automatically created for you and it
contains all the channel variables from that departed channel."

Ok, but how?????

Please help...



From spencer at 5ninesolutions.com  Sun Mar 14 10:49:42 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 10:49:42 -0700
Subject: [Freeswitch-users] 100% CPU
Message-ID: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>

Hi,
I have Freeswitch running as a SBC in front of several Asterisk  
machines and without warning one of the Freeswitch processes will use  
100% of one of the CPU cores and stop responding to SIP requests.  I'm  
using the latest svn on Centos 5.4 in a xen vm.  Basically everything  
will be fine for about 3-4 hours and then this happens.


[root at sip ~]# cat /proc/cpuinfo
processor	: 0
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 0
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 1
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 1
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 2
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 2
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 3
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2204.998
cache size	: 1024 KB
physical id	: 3
siblings	: 1
core id		: 0
cpu cores	: 1
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 apic cmov pat clflush mmx fxsr sse  
sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cmp_legacy
bogomips	: 5514.25
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

[root at sip ~]# uname -r
2.6.18-164.11.1.el5xen


--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980


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From brian at freeswitch.org  Sun Mar 14 10:58:35 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:58:35 -0500
Subject: [Freeswitch-users] js.dll crash on Windows Server 2008 R2 x64
In-Reply-To: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>
References: <736e27571003121303x1df9c407w54f0a717ea16bb53@mail.gmail.com>
Message-ID: <E0BD3C11-9978-4880-8885-0FEB2BE5DEF7@freeswitch.org>

We can't help you because you have failed to provide any information in context.

Please read this link http://wiki.freeswitch.org/wiki/Reporting_Bugs

Then collect the info and report it if the problem persists on latest SVN.

Thanks,
Brain

On Mar 12, 2010, at 3:03 PM, Quentin CALVEZ wrote:

> Hello everybody,
> 
> I've been trying Freeswitch those last days and there is one error that I don't know what to do about. I'm using FusionPBX to manage my freeswitch installation and it works great, however, whenever I try to use a function involving javascript (auto atendants, hunt groups...) the server crashs telling me that the faulty module is js32.dll.
> 
> I really don't know what to do about it. Could someone help me :) ?
> 
> Regards.
> 
> Quentin Calvez
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Mar 14 10:59:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 12:59:36 -0500
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <97634AC6-B936-40F3-9689-CC41E640D0FF@freeswitch.org>

Set the variable sip_cid_type=none and it will only attach the one you set.

/b

On Mar 13, 2010, at 2:38 PM, Roland H?nel wrote:

> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net>
> P-Asserted-Identity: <sip:0123456789 at q-loud.net>

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From brian at freeswitch.org  Sun Mar 14 11:00:10 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:00:10 -0500
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>
References: <50e456911003141040t6022e91av9a139f9ec26fc4b5@mail.gmail.com>
Message-ID: <39B195A1-5A29-418C-A98C-6DCDC829FA39@freeswitch.org>

Pass them as arguments to your luarun hangup hook.

/b

On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:

> Hi guys,
> Freeswitch rocks!
> except the documentation:(
> I'm not able to figure out how to access session variables after hangup...
> I try to dial out on a gateway and detect it the called party hangs up
> before bridge...
> 
> bgapi originate
> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
> 
> I cannot access the variables using:
> 1. script arguments:
> 'luarun hangup.lua ${uuid}'}
> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> 
> 2. session:getVariable("originate_disposition")
> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
> to call global 'getVar' (a nil value)
> 
> 3. params:getHeader("variable_sip_req_uri")
> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
> to index global 'params' (a nil value)
> 
> I have asked this before on the dev list too, but Anthony didn't gave
> me a clean answer here:
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
> 
> "In the script you can access the data from the env event which is
> automatically created for you and it
> contains all the channel variables from that departed channel."
> 
> Ok, but how?????
> 
> Please help...
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Mar 14 11:00:55 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:00:55 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
Message-ID: <A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>

Are you on the latest SVN?

If the problem persists after you update to SVN trunk please read http://wiki.freeswitch.org/wiki/Reporting_Bugs and report the issue to jira.

/b

On Mar 14, 2010, at 12:49 PM, Spencer Thomason wrote:

> I have Freeswitch running as a SBC in front of several Asterisk machines and without warning one of the Freeswitch processes will use 100% of one of the CPU cores and stop responding to SIP requests.  I'm using the latest svn on Centos 5.4 in a xen vm.  Basically everything will be fine for about 3-4 hours and then this happens.
> 




From mrene_lists at avgs.ca  Sun Mar 14 11:02:13 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 14 Mar 2010 14:02:13 -0400
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <C6955D38-7F60-472F-A72F-52C9FAA43595@avgs.ca>

Hi,

If you want to manually add the P-Asserted-Identity header, you must  
first set sip_cid_type to none. Or you could just set  
effective_caller_id_number to the number that should show up in PAI.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 13-Mar-10, at 3:38 PM, Roland H?nel wrote:

> Hello,
>
> I'm developing a very simple call-through application using  
> FreeSWITCH, but currently
> I'm stuck because my provider requires me to send a 'P-Asserted- 
> Identity' header
> for outgoing calls if I want so signal an arbitrary caller ID in the  
> 'From' field.
>
> Setup is as follows:
>    - a-leg comes in from an external gateway
>    - dialplan dispatches the call to a perl handler script
>    - perl script looks like this (simplified as much as possible for  
> this description):
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The  
> Redirector");
>             $session->setVariable("sip_h_P-Asserted-Identity", '<sip:0123456789 at q-loud.net 
> >');
>             $session->execute("bridge", "sofia/gateway/mygateway/ 
> $num);
>
> The problem is now that FreeSWITCH correctly insertes a P-Asserted- 
> Identity header as
> set in the perl script. But there is already an existing P-Asserted- 
> Identity header by default,
> so I end up with two of them. This is the outgoing INVITE (b-leg):
>
> INVITE sip:02222222 at x.x.x.x SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
> Max-Forwards: 63
> From: "The Redirector" <sip:01111111 at q-loud.net>;tag=DBSvZ9NZc750g
> To: <sip:02222222 at x.x.x.x>
> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
> CSeq: 128071597 INVITE
> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 244
> X-Port: 5060
> X-FS-Support: update_display
> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net>
> P-Asserted-Identity: <sip:0123456789 at q-loud.net>
>
> [... sdp ...]
> Numbers are:
>
>     01111111       - the original caller (a-leg & b-leg From:)
>     02222222       - the number redirected to (b-leg To:)
>     0123456789    - the number my provider needs in P-Asserted- 
> Identity ('redirector's number')
>
> So, the INVITE is correct, it includes my P-Asserted-Identity  
> 0123456789
> line, but unfortunately it already includes another P-Asserted- 
> Identity 011111111
> line, and that breaks the setup at my provider.
>
> I really appreciate any help.
>
> Greetings,
> Roland
>
> --
> Roland Haenel
> QSC AG - http://www.qsc.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lawwton at gmail.com  Sun Mar 14 11:06:59 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:06:59 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
Message-ID: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>

All:

We are currently exploring the possibility of using mod_nibblebill and
integrating it with a conference. It seems like it offers all the
features and more that we need. The way we create a conference now is
as follows:

originate sofia/gateway/url/XXXXXXXXXXX
&conference(myConf+flags{moderator,wait-mod})

>From the mod_nibblebill wiki page it seems like we need to set a few
variables for the B leg:

enable_heartbeat_events
nibble_rate
nibble_account=

I was wondering if there is a way to set these variables via the API
call shown above where the conference is created. If not, what options
do you recommend to accomplish this?

Thanks in advance,

Alfredo



From jmesquita at freeswitch.org  Sun Mar 14 11:11:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:11:46 -0300
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
Message-ID: <ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>

Try doing this different. Set the effective_caller_id_number, remove the
manual header insertion and bridge it differently.

Something like this:

            $num = "02222222";
            $session->answer();
            $session->setVariable("effective_caller_id_name", "The
Redirector");
            $session->setVariable("effective_caller_id_number",
"0123456789<sip%3A0123456789 at q-loud.net>
");
            $session->execute("bridge", "{sip_from_uri=
sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
}sofia/gateway/mygateway/$num);


/JM


2010/3/13 Roland H?nel <roland at haenel.me>

> Hello,
>
> I'm developing a very simple call-through application using FreeSWITCH, but
> currently
> I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
> header
> for outgoing calls if I want so signal an arbitrary caller ID in the 'From'
> field.
>
> Setup is as follows:
>    - a-leg comes in from an external gateway
>    - dialplan dispatches the call to a perl handler script
>    - perl script looks like this (simplified as much as possible for this
> description):
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The
> Redirector");
>             $session->setVariable("sip_h_P-Asserted-Identity", '<
> sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
>             $session->execute("bridge", "sofia/gateway/mygateway/$num);
>
> The problem is now that FreeSWITCH correctly insertes a P-Asserted-Identity
> header as
> set in the perl script. But there is already an existing
> P-Asserted-Identity header by default,
> so I end up with two of them. This is the outgoing INVITE (b-leg):
>
>
> INVITE sip:02222222 at x.x.x.x SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
> Max-Forwards: 63
> From: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
> >;tag=DBSvZ9NZc750g
> To: <sip:02222222 at x.x.x.x>
> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
> CSeq: 128071597 INVITE
> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 244
> X-Port: 5060
> X-FS-Support: update_display
> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
> >
> P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
> >
>
> [... sdp ...]
>
> Numbers are:
>
>     01111111       - the original caller (a-leg & b-leg From:)
>     02222222       - the number redirected to (b-leg To:)
>     0123456789    - the number my provider needs in P-Asserted-Identity
> ('redirector's number')
>
> So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
> line, but unfortunately it already includes another P-Asserted-Identity
> 011111111
> line, and that breaks the setup at my provider.
>
> I really appreciate any help.
>
> Greetings,
> Roland
>
> --
> Roland Haenel
> QSC AG - http://www.qsc.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Sun Mar 14 11:14:14 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:14:14 -0300
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
Message-ID: <ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>

Or, I can shut-up and let the ppl who know about this stuff reply to your
email.

JM

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> Try doing this different. Set the effective_caller_id_number, remove the
> manual header insertion and bridge it differently.
>
> Something like this:
>
>             $num = "02222222";
>             $session->answer();
>             $session->setVariable("effective_caller_id_name", "The
> Redirector");
>             $session->setVariable("effective_caller_id_number", "
> 0123456789 <sip%3A0123456789 at q-loud.net>");
>             $session->execute("bridge", "{sip_from_uri=
> sip:01111111 at q-loud.net <sip%3A01111111 at q-loud.net>
> }sofia/gateway/mygateway/$num);
>
>
> /JM
>
>
> 2010/3/13 Roland H?nel <roland at haenel.me>
>
>>  Hello,
>>
>> I'm developing a very simple call-through application using FreeSWITCH,
>> but currently
>> I'm stuck because my provider requires me to send a 'P-Asserted-Identity'
>> header
>> for outgoing calls if I want so signal an arbitrary caller ID in the
>> 'From' field.
>>
>> Setup is as follows:
>>    - a-leg comes in from an external gateway
>>    - dialplan dispatches the call to a perl handler script
>>    - perl script looks like this (simplified as much as possible for this
>> description):
>>
>>             $num = "02222222";
>>             $session->answer();
>>             $session->setVariable("effective_caller_id_name", "The
>> Redirector");
>>             $session->setVariable("sip_h_P-Asserted-Identity", '<
>> sip:0123456789 at q-loud.net <sip%3A0123456789 at q-loud.net>>');
>>             $session->execute("bridge", "sofia/gateway/mygateway/$num);
>>
>> The problem is now that FreeSWITCH correctly insertes a
>> P-Asserted-Identity header as
>> set in the perl script. But there is already an existing
>> P-Asserted-Identity header by default,
>> so I end up with two of them. This is the outgoing INVITE (b-leg):
>>
>>
>> INVITE sip:02222222 at x.x.x.x SIP/2.0
>> Via: SIP/2.0/UDP x.x.x.x:5080;rport;branch=z9hG4bKca3KS0cQj121N
>> Max-Forwards: 63
>> From: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>> >;tag=DBSvZ9NZc750g
>> To: <sip:02222222 at x.x.x.x>
>> Call-ID: ecc19564-a84d-122d-5aa2-00515343ab02
>> CSeq: 128071597 INVITE
>> Contact: <sip:gw+mygateway at x.x.x.x:5080;transport=udp;gw=mygateway>
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16952M
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, refer
>> Privacy: none
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 244
>> X-Port: 5060
>> X-FS-Support: update_display
>> P-Asserted-Identity: "The Redirector" <sip:01111111 at q-loud.net<sip%3A01111111 at q-loud.net>
>> >
>> P-Asserted-Identity: <sip:0123456789 at q-loud.net<sip%3A0123456789 at q-loud.net>
>> >
>>
>> [... sdp ...]
>>
>> Numbers are:
>>
>>     01111111       - the original caller (a-leg & b-leg From:)
>>     02222222       - the number redirected to (b-leg To:)
>>     0123456789    - the number my provider needs in P-Asserted-Identity
>> ('redirector's number')
>>
>> So, the INVITE is correct, it includes my P-Asserted-Identity 0123456789
>> line, but unfortunately it already includes another P-Asserted-Identity
>> 011111111
>> line, and that breaks the setup at my provider.
>>
>> I really appreciate any help.
>>
>> Greetings,
>> Roland
>>
>> --
>> Roland Haenel
>> QSC AG - http://www.qsc.de
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jmesquita at freeswitch.org  Sun Mar 14 11:15:53 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 15:15:53 -0300
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
Message-ID: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>

originate {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>
}sofia/gateway/url/XXXXXXXXXXX
&conference(myConf+flags{moderator,wait-mod})


JM


On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> All:
>
> We are currently exploring the possibility of using mod_nibblebill and
> integrating it with a conference. It seems like it offers all the
> features and more that we need. The way we create a conference now is
> as follows:
>
> originate sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> >From the mod_nibblebill wiki page it seems like we need to set a few
> variables for the B leg:
>
> enable_heartbeat_events
> nibble_rate
> nibble_account=
>
> I was wondering if there is a way to set these variables via the API
> call shown above where the conference is created. If not, what options
> do you recommend to accomplish this?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Sun Mar 14 11:17:27 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 11:17:27 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
Message-ID: <354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>

Yes, I'm using the latest SVN.

Spencer

On Mar 14, 2010, at 11:00 AM, Brian West wrote:

> Are you on the latest SVN?
>
> If the problem persists after you update to SVN trunk please read http://wiki.freeswitch.org/wiki/Reporting_Bugs 
>  and report the issue to jira.
>
> /b
>
> On Mar 14, 2010, at 12:49 PM, Spencer Thomason wrote:
>
>> I have Freeswitch running as a SBC in front of several Asterisk  
>> machines and without warning one of the Freeswitch processes will  
>> use 100% of one of the CPU cores and stop responding to SIP  
>> requests.  I'm using the latest svn on Centos 5.4 in a xen vm.   
>> Basically everything will be fine for about 3-4 hours and then this  
>> happens.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From brian at freeswitch.org  Sun Mar 14 11:20:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 13:20:13 -0500
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
	<ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
Message-ID: <7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>

We are all on like mad today.  :P

/b

On Mar 14, 2010, at 1:14 PM, Jo?o Mesquita wrote:

> Or, I can shut-up and let the ppl who know about this stuff reply to your email.
> 
> JM



From rupa at rupa.com  Sun Mar 14 11:20:45 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 12:20:45 -0600
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
Message-ID: <b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>

Rather than going straight to the conference, you could go through loopback
and have a dialplan that sets the vars and then executes the conference.

On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil <
lawwton at gmail.com> wrote:

> All:
>
> We are currently exploring the possibility of using mod_nibblebill and
> integrating it with a conference. It seems like it offers all the
> features and more that we need. The way we create a conference now is
> as follows:
>
> originate sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> >From the mod_nibblebill wiki page it seems like we need to set a few
> variables for the B leg:
>
> enable_heartbeat_events
> nibble_rate
> nibble_account=
>
> I was wondering if there is a way to set these variables via the API
> call shown above where the conference is created. If not, what options
> do you recommend to accomplish this?
>
> Thanks in advance,
>
> Alfredo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From lloyd.aloysius at gmail.com  Sun Mar 14 11:26:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 14:26:50 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
Message-ID: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>

Hi All,

Can some one help me to install Lua .

OS: CentOS 5.4


wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
tar zxf lua-5.1.4.tar.gz
cd lua-5.1.4
make linux

But getting error when running *make linux *
*
*
*Here is the Output*

In file included from lua.h:16,
                 from lua.c:15:
luaconf.h:275:31: error: readline/readline.h: No such file or directory
luaconf.h:276:30: error: readline/history.h: No such file or directory
lua.c: In function ?pushline?:
lua.c:182: warning: implicit declaration of function ?readline?
lua.c:182: warning: assignment makes pointer from integer without a cast
lua.c: In function ?loadline?:
lua.c:210: warning: implicit declaration of function ?add_history?
make[2]: *** [lua.o] Error 1
make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
make: *** [linux] Error 2


Thank you in advance.


Thanks,
Lloyd
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From lawwton at gmail.com  Sun Mar 14 11:26:55 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:26:55 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
Message-ID: <5fe6fa8f1003141126q5380fec5w3f09a5b13115bb72@mail.gmail.com>

Awesome, thanks a lot.

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>:
> originate
> {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>}sofia/gateway/url/XXXXXXXXXXX
> &conference(myConf+flags{moderator,wait-mod})
>
> JM
>
>
> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From yehavi.bourvine at gmail.com  Sun Mar 14 11:33:48 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 14 Mar 2010 20:33:48 +0200
Subject: [Freeswitch-users] Call transfer with Polycom works ok but a
	call is left on hold?
In-Reply-To: <C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>
References: <d361ca7b1003140818x26ea86efi8a6582a798751def@mail.gmail.com>
	<C40AAEE1-E04A-4417-8C2E-27D39727B80C@freeswitch.org>
Message-ID: <d361ca7b1003141133t5ad5d74ct9f27b99322dc361d@mail.gmail.com>

I am using the almost latest SVN (from yesterday) and our custom dialplan.
I'll try stripping down our dialplan to see whether it helps.

                     Thanks, __Yehavi:

2010/3/14 Brian West <brian at freeswitch.org>

> Are you doing this with the latest SVN and default dialplan or are you
> doing something out of the usual?
>
> /b
>
> On Mar 14, 2010, at 10:18 AM, Yehavi Bourvine wrote:
>
> > Hello,
> >
> >    We have a strange behavior which my users told me today, but I have a
> feeling it lasts for a long time (we are running now version 16972).
> >
> > User A calls user B on a Polycom. User B press the "transfer" key and
> calls C who answers. B then press "transfer" again, and now A talks to C.
> However, B is left with a held call to the voicemail of C...
> >
> > Anyone saw this behaviour?  unattended transfer and conference work ok.
> >
> >                              Thanks! __Yehavi:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lawwton at gmail.com  Sun Mar 14 11:38:35 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:38:35 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
Message-ID: <5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>

Thanks Rupa, We have looked at the loopback as an option and really
like the idea, the only thing there is that the wiki page says:

"Please take extra precautions in regard to CDR records. They may not
display the information you are looking for after a call has been
bridged through the loopback endpoint."

We are not quite sure what that statement translates to as far as what
fields from the CDR will be affected, etc...

On Sun, Mar 14, 2010 at 2:20 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> Rather than going straight to the conference, you could go through loopback
> and have a dialplan that sets the vars and then executes the conference.
>
> On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil
> <lawwton at gmail.com> wrote:
>>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From nazim.agabekov at gmail.com  Sun Mar 14 11:38:45 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Sun, 14 Mar 2010 22:38:45 +0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
Message-ID: <4B9D2D35.8010303@gmail.com>

Hi Aloysius,

Probably readline development package is not installed. If you have
difficulties compiling Lua,
you could get binaries + includes from:
http://luabinaries.luaforge.net/download.html

On 03/14/2010 10:26 PM, Aloysius Lloyd wrote:
> Hi All,
>
> Can some one help me to install Lua . 
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running *make linux *
> *
> *
> *Here is the Output*
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From rupa at rupa.com  Sun Mar 14 11:40:16 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 12:40:16 -0600
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
Message-ID: <b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>

That will set it on the a-leg.  IN this case it probably doesn't matter and
it is cleaner to set it on the a-leg.

2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> originate {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>
> }sofia/gateway/url/XXXXXXXXXXX
>
> &conference(myConf+flags{moderator,wait-mod})
>
>
> JM
>
>
>
> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil <
> lawwton at gmail.com> wrote:
>
>> All:
>>
>> We are currently exploring the possibility of using mod_nibblebill and
>> integrating it with a conference. It seems like it offers all the
>> features and more that we need. The way we create a conference now is
>> as follows:
>>
>> originate sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> >From the mod_nibblebill wiki page it seems like we need to set a few
>> variables for the B leg:
>>
>> enable_heartbeat_events
>> nibble_rate
>> nibble_account=
>>
>> I was wondering if there is a way to set these variables via the API
>> call shown above where the conference is created. If not, what options
>> do you recommend to accomplish this?
>>
>> Thanks in advance,
>>
>> Alfredo
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From spencer at 5ninesolutions.com  Sun Mar 14 11:40:17 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 11:40:17 -0700
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
Message-ID: <8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>

I have Centos RPMS and source RPMS for lua 5.1.4 at http://repo.5ninesolutions.com
--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980



On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:

> Hi All,
>
> Can some one help me to install Lua .
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running make linux
>
> Here is the Output
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or  
> directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a  
> cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rm at callrica.co.za  Sun Mar 14 11:41:48 2010
From: rm at callrica.co.za (Roly Maz)
Date: Sun, 14 Mar 2010 20:41:48 +0200
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
Message-ID: <028601cac3a6$056fc940$104f5bc0$@co.za>

Hi Bruce/Brian, 

 

Success on my side too! 

 

Thanks for sharing your experience. I too reset my FS and made minimal
changes to my config. Voila! 

 

Anyone new to FS, and running it on Windows server- I strongly recommend you
pull a copy of Wireshark off the net. It has great tools for analysing
SIP/IP packets. 

 

I'll be adding my config and service provider setup to the wiki this week. 

 

 

Rgds

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 14 March 2010 12:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

 


Hi Roly,

I promised I'd let you know when I got it working - so here it is:

As Brian said, doublenat is no longer needed.

It looks like I must have just been confusing things horribly by using
various things I'd read about, including:

<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

<param name="aggressive-nat-detection" value="true"/>

 

These, and who knows what other mucking about I had been doing, must have
confused everything as none of this is now necessary.

 

All I did - after a reinstall - to get everything working through double
NATs is:

 

1.  In vars.xml 

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>

 

<X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>

 

2.  Then in prefix/sip_profiles/internal.xml

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<param name="ext-sip-ip" value="$${external_sip_ip)"/>

 

Then as the man says, it just works.  

I decided not to enable options pings to keep the far-end NAT open to enable
calling the remote phone - instead I configured it to send keep-alives.

 

And that's it.  I can't believe the knots I was tying myself in.  It just
works !

cheers

Bruce

 

 

 

 

 

 

On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:

Ah I see... what? Please share and lead this blind man out the FS
wilderness!

I don't understand...what happens to the external profile? Do you delete it?
And how do you forward port 5060?

...and you thought you were a newbie!

Any insight would be much appreciated...loving the journey.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
Hopkins
Sent: 09 March 2010 08:23 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] doublenat outgoing call issue

Ah I see.  I will try again using the internal profile and forwarding
port 5060.  Presumably still creating a directory entry to enable the
outside-facing domain to be used.

Many thanks for your patient help of a newbie Brian.

Cheers
Bruce

Please excuse the brevity - sent from my mobile.

On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:

> ok you aren't catching one key thing here.. you no longer need two
> profiles.
>
> /b
>
> On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
>
>> Hi again,
>>
>> Thanks for the help so far.  Unfotrunately I must still be doing
>> something wrong here as I am still having difficulty, and still
>> have the same problem.
>>
>> I updated to build 16938 by means of "make current"
>>
>> I'm not able to use UPNP or NATPMP so changed the port forwarding
>> to use 5080 instead of 5090.
>>
>> I got rid of the doublenat profile in sip_profiles, though I had to
>> retain an entry in the directory /usr/local/freeeswitch/conf/
>> directory/ext_dns.xml in order to give freeswitch the dns name of
>> the server as a domain for the remote softphone to register on.  I
>> left the group name in this entry the same as inthe default entry,
>> so that the remote phone could register on the same extension
>> numbers (100, etc) as in the default build.
>>
>> I still find that, if I initiate a call from the local (on same LAN
>> as freeswitch) phone to the remote phone, I get the message on the
>> CLI:
>>
>> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
>> type [user] cause: [USER_NOT_REGISTERED]
>>
>> One possibly unrelated aside, I also found I needed to uncomment
>> <param name="aggressive-nat-detection" value="true"/> in
>> external.xml, otherwise in the case of a call initiated by the
>> remote phone being hung up by the local phone, freeswitch sent the
>> BYE to the private IP of the remote phone, rather than its public
>> ip - meaning that the remote phone didn't receive the BYE.
>>
>> Any further ideas where I am going wrong here please?
>>
>> thanks again in advance
>> Bruce
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From lloyd.aloysius at gmail.com  Sun Mar 14 11:49:27 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 14:49:27 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
Message-ID: <8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>

Nazim & Spencer , thank you.

readline development package install solved the problem. Here is the Lua
install on CentOS 5.4

yum install readline-devel
wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
tar zxf lua-5.1.4.tar.gz
cd lua-5.1.4
make linux
make linux install

Thanks
Lloyd



On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I have Centos RPMS and source RPMS for lua 5.1.4 at
> http://repo.5ninesolutions.com
>  --
> *Spencer Thomason*
> 5Nine Solutions LLC
> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
> p. +1.888.271.7959
> f. +1.310.510.6980
>
>  <http://www.5ninesolutions.com/>
>
> On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>
> Hi All,
>
> Can some one help me to install Lua .
>
> OS: CentOS 5.4
>
>
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
>
> But getting error when running *make linux *
> *
> *
> *Here is the Output*
>
> In file included from lua.h:16,
>                  from lua.c:15:
> luaconf.h:275:31: error: readline/readline.h: No such file or directory
> luaconf.h:276:30: error: readline/history.h: No such file or directory
> lua.c: In function ?pushline?:
> lua.c:182: warning: implicit declaration of function ?readline?
> lua.c:182: warning: assignment makes pointer from integer without a cast
> lua.c: In function ?loadline?:
> lua.c:210: warning: implicit declaration of function ?add_history?
> make[2]: *** [lua.o] Error 1
> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
> make[1]: *** [linux] Error 2
> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
> make: *** [linux] Error 2
>
>
> Thank you in advance.
>
>
> Thanks,
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lawwton at gmail.com  Sun Mar 14 11:50:23 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Sun, 14 Mar 2010 14:50:23 -0400
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<ebbc2d551003141115v5816626evf0648ddd1d04c342@mail.gmail.com>
	<b39273531003141140t333e5d64u77cb3dff981c7dfb@mail.gmail.com>
Message-ID: <5fe6fa8f1003141150u41d60e71u2d1446d00950efdf@mail.gmail.com>

Thanks Rupa, appreciate the help.

On Sun, Mar 14, 2010 at 2:40 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> That will set it on the a-leg. ?IN this case it probably doesn't matter and
> it is cleaner to set it on the a-leg.
>
> 2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> originate
>> {enable_heartbeat_events,nibble_rate=<RATE>,nibble_account=<NUM>}sofia/gateway/url/XXXXXXXXXXX
>> &conference(myConf+flags{moderator,wait-mod})
>>
>> JM
>>
>>
>> On Sun, Mar 14, 2010 at 3:06 PM, Alfredo Quiroga-Villamil
>> <lawwton at gmail.com> wrote:
>>>
>>> All:
>>>
>>> We are currently exploring the possibility of using mod_nibblebill and
>>> integrating it with a conference. It seems like it offers all the
>>> features and more that we need. The way we create a conference now is
>>> as follows:
>>>
>>> originate sofia/gateway/url/XXXXXXXXXXX
>>> &conference(myConf+flags{moderator,wait-mod})
>>>
>>> >From the mod_nibblebill wiki page it seems like we need to set a few
>>> variables for the B leg:
>>>
>>> enable_heartbeat_events
>>> nibble_rate
>>> nibble_account=
>>>
>>> I was wondering if there is a way to set these variables via the API
>>> call shown above where the conference is created. If not, what options
>>> do you recommend to accomplish this?
>>>
>>> Thanks in advance,
>>>
>>> Alfredo
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From lloyd.aloysius at gmail.com  Sun Mar 14 12:46:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 14 Mar 2010 15:46:46 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
Message-ID: <8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>

I am trying to install Luasql.  Lots of make errors.Here is the steps I
follow

wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
tar xfvz luasql-2.1.1.tar.gz
cd luasql-2.1.1/
vi config

uncomment the mysql
enable mysql and change the mysql lib and include directory path.


How to install LuaSql in CentOS 5.4?

What is the best way to access mysql from Lua?
odbc or natvie?

Thanks
Lloyd


On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Nazim & Spencer , thank you.
>
> readline development package install solved the problem. Here is the Lua
> install on CentOS 5.4
>
> yum install readline-devel
> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
> tar zxf lua-5.1.4.tar.gz
> cd lua-5.1.4
> make linux
> make linux install
>
> Thanks
> Lloyd
>
>
>
> On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I have Centos RPMS and source RPMS for lua 5.1.4 at
>> http://repo.5ninesolutions.com
>>  --
>> *Spencer Thomason*
>> 5Nine Solutions LLC
>> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
>> p. +1.888.271.7959
>> f. +1.310.510.6980
>>
>>  <http://www.5ninesolutions.com/>
>>
>> On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>>
>> Hi All,
>>
>> Can some one help me to install Lua .
>>
>> OS: CentOS 5.4
>>
>>
>> wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>> tar zxf lua-5.1.4.tar.gz
>> cd lua-5.1.4
>> make linux
>>
>> But getting error when running *make linux *
>> *
>> *
>> *Here is the Output*
>>
>> In file included from lua.h:16,
>>                  from lua.c:15:
>> luaconf.h:275:31: error: readline/readline.h: No such file or directory
>> luaconf.h:276:30: error: readline/history.h: No such file or directory
>> lua.c: In function ?pushline?:
>> lua.c:182: warning: implicit declaration of function ?readline?
>> lua.c:182: warning: assignment makes pointer from integer without a cast
>> lua.c: In function ?loadline?:
>> lua.c:210: warning: implicit declaration of function ?add_history?
>> make[2]: *** [lua.o] Error 1
>> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>> make[1]: *** [linux] Error 2
>> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>> make: *** [linux] Error 2
>>
>>
>> Thank you in advance.
>>
>>
>> Thanks,
>> Lloyd
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From rentmycoder at gmail.com  Sun Mar 14 13:01:54 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Sun, 14 Mar 2010 21:01:54 +0100
Subject: [Freeswitch-users] how to access session variables after
	hangup???
Message-ID: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>

> Pass them as arguments to your luarun hangup hook.
And what about variables are not available before the call setup (like
uuid), only during the call progress or after the hangup???
I would need to access the originate_disposition variable...

>> 'luarun hangup.lua ${originate_disposition}'
>> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data (${api_hangup_hook} contains a variable)
any other hint???


> ---------- Forwarded message ----------
> From:?Brian West <brian at freeswitch.org>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Sun, 14 Mar 2010 13:00:10 -0500
> Subject:?Re: [Freeswitch-users] how to access session variables after hangup???
> Pass them as arguments to your luarun hangup hook.
>
> /b
>
> On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:
>
>> Hi guys,
>> Freeswitch rocks!
>> except the documentation:(
>> I'm not able to figure out how to access session variables after hangup...
>> I try to dial out on a gateway and detect it the called party hangs up
>> before bridge...
>>
>> bgapi originate
>> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
>> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
>>
>> I cannot access the variables using:
>> 1. script arguments:
>> 'luarun hangup.lua ${uuid}'}
>> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
>> (${api_hangup_hook} contains a variable)
>>
>> 2. session:getVariable("originate_disposition")
>> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
>> to call global 'getVar' (a nil value)
>>
>> 3. params:getHeader("variable_sip_req_uri")
>> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
>> to index global 'params' (a nil value)
>>
>> I have asked this before on the dev list too, but Anthony didn't gave
>> me a clean answer here:
>> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
>>
>> "In the script you can access the data from the env event which is
>> automatically created for you and it
>> contains all the channel variables from that departed channel."
>>
>> Ok, but how?????
>>
>> Please help...
>>



From spencer at 5ninesolutions.com  Sun Mar 14 13:13:16 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 13:13:16 -0700
Subject: [Freeswitch-users] Dial number at registered extension
Message-ID: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>

Hello all,
Is there a way I can dial at number at a registered extension.   
Basically what I'd like to do is use Freeswitch as a SBC for a number  
of asterisk machines and I'd like to pass an incoming sip invite to a  
specific number at each one.  Right now I'm specifying the IP in the  
bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if  
I dial the registered user directly it dials the contact address at  
the registered user and does not pass the inbound number.  Basically  
what I'm trying to do is this:

pstn -> ITSP -> freeswitch -> asterisk (1-n)

Thanks,
Spencer



From roland at haenel.me  Sun Mar 14 13:45:12 2010
From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=)
Date: Sun, 14 Mar 2010 21:45:12 +0100
Subject: [Freeswitch-users] Trouble with P-Asserted-Identity header(s)
In-Reply-To: <7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>
References: <429591151003131238q1d1531erbefc732b8e813126@mail.gmail.com>
	<ebbc2d551003141111y77b1a99ck860565e3385dfdc5@mail.gmail.com>
	<ebbc2d551003141114q53deeb25t6b21ed7e91b75525@mail.gmail.com>
	<7C5C714B-C507-4555-9153-2CF1BA9E8192@freeswitch.org>
Message-ID: <429591151003141345n67639652g9fffbafa689c748d@mail.gmail.com>

Thanks guys for being mad like this.
Not only gave me the perfect solution but also some insight about
FreeSWITCH. :-)

Greetings,
Roland
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From nazim.agabekov at gmail.com  Sun Mar 14 13:59:24 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Mon, 15 Mar 2010 00:59:24 +0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
	<8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
Message-ID: <4B9D4E2C.4020407@gmail.com>

I would not use luasql with mysql driver because it leaks memory. Luasql
odbc is better choice for production.
What kind of errors do you have? If you installing luasql.odbc please
make sure you have unix-odbc development packages.

Nazim




On 03/14/2010 11:46 PM, Aloysius Lloyd wrote:
> I am trying to install Luasql.  Lots of make errors.Here is the steps
> I follow
>
> wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
> tar xfvz luasql-2.1.1.tar.gz
> cd luasql-2.1.1/
> vi config
>
> uncomment the mysql
> enable mysql and change the mysql lib and include directory path.
>
>
> How to install LuaSql in CentOS 5.4?
>
> What is the best way to access mysql from Lua? 
> odbc or natvie?
>
> Thanks
> Lloyd
>
>
> On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd
> <lloyd.aloysius at gmail.com <mailto:lloyd.aloysius at gmail.com>> wrote:
>
>     Nazim & Spencer , thank you.
>
>     readline development package install solved the problem. Here is
>     the Lua install on CentOS 5.4
>
>     yum install readline-devel
>     wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>     tar zxf lua-5.1.4.tar.gz
>     cd lua-5.1.4
>     make linux 
>     make linux install
>
>     Thanks
>     Lloyd
>
>
>
>     On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason
>     <spencer at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>>
>     wrote:
>
>         I have Centos RPMS and source RPMS for lua 5.1.4 at
>         http://repo.5ninesolutions.com
>         --
>         *Spencer Thomason*
>         5Nine Solutions LLC
>         e. sales at 5ninesolutions.com <mailto:spencer at 5ninesolutions.com>
>         p. +1.888.271.7959
>         f. +1.310.510.6980
>
>         <http://www.5ninesolutions.com/>
>
>         On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>
>>         Hi All,
>>
>>         Can some one help me to install Lua . 
>>
>>         OS: CentOS 5.4
>>
>>
>>         wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>>         tar zxf lua-5.1.4.tar.gz
>>         cd lua-5.1.4
>>         make linux
>>
>>         But getting error when running *make linux *
>>         *
>>         *
>>         *Here is the Output*
>>
>>         In file included from lua.h:16,
>>                          from lua.c:15:
>>         luaconf.h:275:31: error: readline/readline.h: No such file or
>>         directory
>>         luaconf.h:276:30: error: readline/history.h: No such file or
>>         directory
>>         lua.c: In function ?pushline?:
>>         lua.c:182: warning: implicit declaration of function ?readline?
>>         lua.c:182: warning: assignment makes pointer from integer
>>         without a cast
>>         lua.c: In function ?loadline?:
>>         lua.c:210: warning: implicit declaration of function
>>         ?add_history?
>>         make[2]: *** [lua.o] Error 1
>>         make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>>         make[1]: *** [linux] Error 2
>>         make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>>         make: *** [linux] Error 2
>>
>>
>>         Thank you in advance.
>>
>>
>>         Thanks,
>>         Lloyd
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From rupa at rupa.com  Sun Mar 14 14:10:53 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 16:10:53 -0500
Subject: [Freeswitch-users] mod_nibblebill and dynamic conference
In-Reply-To: <5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>
References: <5fe6fa8f1003141106q462a3988k59d192f92cbd441@mail.gmail.com>
	<b39273531003141120y64e6db42h59492eb024d0f9f1@mail.gmail.com>
	<5fe6fa8f1003141138g5d9010e2h826053ad2aba7a44@mail.gmail.com>
Message-ID: <b39273531003141410q12a580b3s8c8dd781eb3f66d7@mail.gmail.com>

Basically you'll get new CDR records for the loopback side of the call. So
you may have to reconcile the two -- depends on how you process your CDRs.
 If you can avoid loopback, do so.  In this case, doing nibblebill on the
a-leg is the right way to go unless you have a specific reason to nibble on
the b-leg.

On Sun, Mar 14, 2010 at 1:38 PM, Alfredo Quiroga-Villamil <lawwton at gmail.com
> wrote:

> Thanks Rupa, We have looked at the loopback as an option and really
> like the idea, the only thing there is that the wiki page says:
>
> "Please take extra precautions in regard to CDR records. They may not
> display the information you are looking for after a call has been
> bridged through the loopback endpoint."
>
> We are not quite sure what that statement translates to as far as what
> fields from the CDR will be affected, etc...
>
> On Sun, Mar 14, 2010 at 2:20 PM, Rupa Schomaker <rupa at rupa.com> wrote:
> > Rather than going straight to the conference, you could go through
> loopback
> > and have a dialplan that sets the vars and then executes the conference.
> >
> > On Sun, Mar 14, 2010 at 12:06 PM, Alfredo Quiroga-Villamil
> > <lawwton at gmail.com> wrote:
> >>
> >> All:
> >>
> >> We are currently exploring the possibility of using mod_nibblebill and
> >> integrating it with a conference. It seems like it offers all the
> >> features and more that we need. The way we create a conference now is
> >> as follows:
> >>
> >> originate sofia/gateway/url/XXXXXXXXXXX
> >> &conference(myConf+flags{moderator,wait-mod})
> >>
> >> >From the mod_nibblebill wiki page it seems like we need to set a few
> >> variables for the B leg:
> >>
> >> enable_heartbeat_events
> >> nibble_rate
> >> nibble_account=
> >>
> >> I was wondering if there is a way to set these variables via the API
> >> call shown above where the conference is created. If not, what options
> >> do you recommend to accomplish this?
> >>
> >> Thanks in advance,
> >>
> >> Alfredo
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > -Rupa
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Sun Mar 14 14:12:28 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Sun, 14 Mar 2010 16:12:28 -0500
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
Message-ID: <b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>

maybe look at using something like mod_easyroute.  You setup a database of
mappings between DID and endpoints.  It is designed for the situation you
describe.

On Sun, Mar 14, 2010 at 3:13 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> Is there a way I can dial at number at a registered extension.
> Basically what I'd like to do is use Freeswitch as a SBC for a number
> of asterisk machines and I'd like to pass an incoming sip invite to a
> specific number at each one.  Right now I'm specifying the IP in the
> bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if
> I dial the registered user directly it dials the contact address at
> the registered user and does not pass the inbound number.  Basically
> what I'm trying to do is this:
>
> pstn -> ITSP -> freeswitch -> asterisk (1-n)
>
> Thanks,
> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From spencer at 5ninesolutions.com  Sun Mar 14 15:02:33 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 15:02:33 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
Message-ID: <AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>

That looks promising, thanks.  I was hoping to have FS do it  
automatically.  I.e. inbound dids are assigned to specific asterisk  
boxes, ITSP dials inbound did of freeswitch, freeswitch dials inbound  
did @ whereever the registration came from instead of having the IP  
address hard coded into a route.


On Mar 14, 2010, at 2:12 PM, Rupa Schomaker wrote:

> maybe look at using something like mod_easyroute.  You setup a  
> database of mappings between DID and endpoints.  It is designed for  
> the situation you describe.
>
> On Sun, Mar 14, 2010 at 3:13 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> Hello all,
> Is there a way I can dial at number at a registered extension.
> Basically what I'd like to do is use Freeswitch as a SBC for a number
> of asterisk machines and I'd like to pass an incoming sip invite to a
> specific number at each one.  Right now I'm specifying the IP in the
> bridge command like this sofia/default/NNNNNNNNNNN at xx.xx.xx.xx but if
> I dial the registered user directly it dials the contact address at
> the registered user and does not pass the inbound number.  Basically
> what I'm trying to do is this:
>
> pstn -> ITSP -> freeswitch -> asterisk (1-n)
>
> Thanks,
> Spencer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> -Rupa
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Mar 14 15:18:05 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 17:18:05 -0500
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
Message-ID: <2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>

You sure can still have it call a registration.. You're mixing the dialplan and user directly as if they are a single entity.  They aren't.

/b

On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:

> That looks promising, thanks.  I was hoping to have FS do it automatically.  I.e. inbound dids are assigned to specific asterisk boxes, ITSP dials inbound did of freeswitch, freeswitch dials inbound did @ whereever the registration came from instead of having the IP address hard coded into a route.




From spencer at 5ninesolutions.com  Sun Mar 14 16:00:26 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 16:00:26 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
Message-ID: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>

I guess my question is, is there some function which will return the  
IP address of a registered user?  Similar to sofia_contact but only  
the IP address?

Spencer

On Mar 14, 2010, at 3:18 PM, Brian West wrote:

> You sure can still have it call a registration.. You're mixing the  
> dialplan and user directly as if they are a single entity.  They  
> aren't.
>
> /b
>
> On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>
>> That looks promising, thanks.  I was hoping to have FS do it  
>> automatically.  I.e. inbound dids are assigned to specific asterisk  
>> boxes, ITSP dials inbound did of freeswitch, freeswitch dials  
>> inbound did @ whereever the registration came from instead of  
>> having the IP address hard coded into a route.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From anthony.minessale at gmail.com  Sun Mar 14 16:08:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Mar 2010 17:08:47 -0600
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
Message-ID: <191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>

Do you think its due to resource limitations in your vm?
Make sure you check all the config options to give fs full access to all the
resouces.

If it persists, do top -H and identify the pid with the 100% usage.  Then
get a gcore and gdb it to capture the output of "thread apply all bt"

Report the findings to jira.

On Mar 14, 2010 1:23 PM, "Spencer Thomason" <spencer at 5ninesolutions.com>
wrote:

Yes, I'm using the latest SVN.

Spencer


On Mar 14, 2010, at 11:00 AM, Brian West wrote:

> Are you on the latest SVN?
>
> If the problem pe...
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From jmesquita at freeswitch.org  Sun Mar 14 16:17:13 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 20:17:13 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
Message-ID: <ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>

Yes, sofia_contact with regex to extract the IP portion? You had the answer
yourself.

Jo?o Mesquita



On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Sun Mar 14 16:17:36 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 20:17:36 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
Message-ID: <ebbc2d551003141617n12b2f81djfba1e6d6a7b8fe66@mail.gmail.com>

Oh, check the RURI example on default configs. That might give you a hint.

Jo?o Mesquita



On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From spencer at 5ninesolutions.com  Sun Mar 14 17:14:11 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:14:11 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
Message-ID: <E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>

No.  Its on our hardware, we just use xen to virtualize the machines..  
this is the only vm on the machine.  Gcore complains about no  
debugging symbols found, do I need to rebuild?

It appears to be a problem when registering to one of my trunks.  The  
very strange thing is the everything will be working perfectly and  
then all of a sudden this happens, i.e. trunks regged, calls  
completing, etc. This is a very lightly loaded server.

This is the output from the console:

freeswitch at internal> 2010-03-14 23:12:15.199983 [NOTICE] sofia_reg.c: 
336 Registering bandtel
2010-03-14 23:12:15.223977 [DEBUG] sofia_reg.c:1478 Changing expire  
time to 475 by request of proxy sip:66.237.65.67
2010-03-14 23:20:05.429613 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:20:08.285756 [ERR] sofia_reg.c:1490 bandtel Registration  
Failed with status Operation has no matching challenge  [904]. failure  
#1
2010-03-14 23:20:10.043933 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:21:16.332797 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:21:52.528409 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:22:56.945511 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:23:33.083177 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:24:37.398155 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:25:13.631825 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:26:17.978767 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:26:54.172425 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.
2010-03-14 23:27:58.579354 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-14 23:28:34.605017 [WARNING] sofia_reg.c:381 bandtel Failed  
Registration, setting retry to 60 seconds.


On Mar 14, 2010, at 4:08 PM, Anthony Minessale wrote:

> Do you think its due to resource limitations in your vm?
> Make sure you check all the config options to give fs full access to  
> all the resouces.
>
> If it persists, do top -H and identify the pid with the 100% usage.   
> Then get a gcore and gdb it to capture the output of "thread apply  
> all bt"
>
> Report the findings to jira.
>
>> On Mar 14, 2010 1:23 PM, "Spencer Thomason" <spencer at 5ninesolutions.com 
>> > wrote:
>>
>> Yes, I'm using the latest SVN.
>>
>> Spencer
>>
>> On Mar 14, 2010, at 11:00 AM, Brian West wrote:
>>
>> > Are you on the latest SVN?
>> >
>> > If the problem pe...
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Mar 14 17:23:43 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 19:23:43 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
Message-ID: <F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>

Are you sure you're on SVN trunk?

/b

On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:

> No.  Its on our hardware, we just use xen to virtualize the machines.. this is the only vm on the machine.  Gcore complains about no debugging symbols found, do I need to rebuild?  
> 
> It appears to be a problem when registering to one of my trunks.  The very strange thing is the everything will be working perfectly and then all of a sudden this happens, i.e. trunks regged, calls completing, etc. This is a very lightly loaded server.
> 
> This is the output from the console:




From spencer at 5ninesolutions.com  Sun Mar 14 17:25:43 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:25:43 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
Message-ID: <10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>

Duh.. :-)  Thanks!

On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:

> Yes, sofia_contact with regex to extract the IP portion? You had the  
> answer yourself.
>
> Jo?o Mesquita
>
>
>
> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> I guess my question is, is there some function which will return the
> IP address of a registered user?  Similar to sofia_contact but only
> the IP address?
>
> Spencer
>
> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>
> > You sure can still have it call a registration.. You're mixing the
> > dialplan and user directly as if they are a single entity.  They
> > aren't.
> >
> > /b
> >
> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
> >
> >> That looks promising, thanks.  I was hoping to have FS do it
> >> automatically.  I.e. inbound dids are assigned to specific asterisk
> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
> >> inbound did @ whereever the registration came from instead of
> >> having the IP address hard coded into a route.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From spencer at 5ninesolutions.com  Sun Mar 14 17:38:43 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 17:38:43 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
Message-ID: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>

I rebuilt today.

On Mar 14, 2010, at 5:23 PM, Brian West wrote:

> Are you sure you're on SVN trunk?
>
> /b
>
> On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
>
>> No.  Its on our hardware, we just use xen to virtualize the  
>> machines.. this is the only vm on the machine.  Gcore complains  
>> about no debugging symbols found, do I need to rebuild?
>>
>> It appears to be a problem when registering to one of my trunks.   
>> The very strange thing is the everything will be working perfectly  
>> and then all of a sudden this happens, i.e. trunks regged, calls  
>> completing, etc. This is a very lightly loaded server.
>>
>> This is the output from the console:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From brian at freeswitch.org  Sun Mar 14 17:44:36 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 19:44:36 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
Message-ID: <C6CA5510-F725-4612-B39C-AF0E959A1E3E@freeswitch.org>

Collect sip traces for that issue.

/b

On Mar 14, 2010, at 7:38 PM, Spencer Thomason wrote:

> I rebuilt today.
> 
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
> 
>> Are you sure you're on SVN trunk?
>> 
>> /b




From jbrucehopkins at gmail.com  Sun Mar 14 18:53:10 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 15 Mar 2010 01:53:10 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <028601cac3a6$056fc940$104f5bc0$@co.za>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
Message-ID: <d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>

Doubles all round !

cheers
Bruce

On 14 March 2010 18:41, Roly Maz <rm at callrica.co.za> wrote:

>  Hi Bruce/Brian,
>
>
>
> Success on my side too!
>
>
>
> Thanks for sharing your experience. I too reset my FS and made minimal
> changes to my config. Voila!
>
>
>
> Anyone new to FS, and running it on Windows server? I strongly recommend
> you pull a copy of Wireshark off the net. It has great tools for analysing
> SIP/IP packets.
>
>
>
> I?ll be adding my config and service provider setup to the wiki this week.
>
>
>
>
>
> Rgds
>
>
>
>
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bruce
> Hopkins
> *Sent:* 14 March 2010 12:57 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] doublenat outgoing call issue
>
>
>
>
> Hi Roly,
>
> I promised I'd let you know when I got it working - so here it is:
>
> As Brian said, doublenat is no longer needed.
>
> It looks like I must have just been confusing things horribly by using
> various things I'd read about, including:
>
> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
>
> <param name="aggressive-nat-detection" value="true"/>
>
>
>
> These, and who knows what other mucking about I had been doing, must have
> confused everything as none of this is now necessary.
>
>
>
> All I did - after a reinstall - to get everything working through double
> NATs is:
>
>
>
> 1.  In vars.xml
>
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=<my.fs.ext.ip>"/>
>
>
>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=<my.fs.ext.ip>"/>
>
>
>
> 2.  Then in prefix/sip_profiles/internal.xml
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_sip_ip)"/>
>
>
>
> Then as the man says, it just works.
>
> I decided not to enable options pings to keep the far-end NAT open to
> enable calling the remote phone - instead I configured it to send
> keep-alives.
>
>
>
> And that's it.  I can't believe the knots I was tying myself in.  It just
> works !
>
> cheers
>
> Bruce
>
>
>
>
>
>
>
>
>
>
>
>
>
> On 9 March 2010 19:08, Roly Maz <rm at callrica.co.za> wrote:
>
> Ah I see... what? Please share and lead this blind man out the FS
> wilderness!
>
> I don't understand...what happens to the external profile? Do you delete
> it?
> And how do you forward port 5060?
>
> ...and you thought you were a newbie!
>
> Any insight would be much appreciated...loving the journey.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bruce
> Hopkins
> Sent: 09 March 2010 08:23 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] doublenat outgoing call issue
>
> Ah I see.  I will try again using the internal profile and forwarding
> port 5060.  Presumably still creating a directory entry to enable the
> outside-facing domain to be used.
>
> Many thanks for your patient help of a newbie Brian.
>
> Cheers
> Bruce
>
> Please excuse the brevity - sent from my mobile.
>
> On 8 Mar 2010, at 17:21, Brian West <brian at freeswitch.org> wrote:
>
> > ok you aren't catching one key thing here.. you no longer need two
> > profiles.
> >
> > /b
> >
> > On Mar 8, 2010, at 11:12 AM, Bruce Hopkins wrote:
> >
> >> Hi again,
> >>
> >> Thanks for the help so far.  Unfotrunately I must still be doing
> >> something wrong here as I am still having difficulty, and still
> >> have the same problem.
> >>
> >> I updated to build 16938 by means of "make current"
> >>
> >> I'm not able to use UPNP or NATPMP so changed the port forwarding
> >> to use 5080 instead of 5090.
> >>
> >> I got rid of the doublenat profile in sip_profiles, though I had to
> >> retain an entry in the directory /usr/local/freeeswitch/conf/
> >> directory/ext_dns.xml in order to give freeswitch the dns name of
> >> the server as a domain for the remote softphone to register on.  I
> >> left the group name in this entry the same as inthe default entry,
> >> so that the remote phone could register on the same extension
> >> numbers (100, etc) as in the default build.
> >>
> >> I still find that, if I initiate a call from the local (on same LAN
> >> as freeswitch) phone to the remote phone, I get the message on the
> >> CLI:
> >>
> >> [ERR] switch_ivr_originate.c2389 Cannot create outgoing channel of
> >> type [user] cause: [USER_NOT_REGISTERED]
> >>
> >> One possibly unrelated aside, I also found I needed to uncomment
> >> <param name="aggressive-nat-detection" value="true"/> in
> >> external.xml, otherwise in the case of a call initiated by the
> >> remote phone being hung up by the local phone, freeswitch sent the
> >> BYE to the private IP of the remote phone, rather than its public
> >> ip - meaning that the remote phone didn't receive the BYE.
> >>
> >> Any further ideas where I am going wrong here please?
> >>
> >> thanks again in advance
> >> Bruce
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Mar 14 19:06:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 14 Mar 2010 21:06:15 -0500
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
References: <d03a10e01003080456p69a6f2e1yd1b7196cf78a48be@mail.gmail.com>
	<4AD66A73-DC6B-461E-A752-C44D60ADA243@freeswitch.org>
	<d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
Message-ID: <A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>

Its nice to hear more people have this working.  The more that depend on this .. the simpler my job is... because If we break it they start to open jira's about it.

/b

On Mar 14, 2010, at 8:53 PM, Bruce Hopkins wrote:

> Doubles all round !
> 
> cheers
> Bruce




From jason at jasonjgw.net  Sun Mar 14 19:17:46 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Mar 2010 13:17:46 +1100
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
References: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<C9316851-99DC-4989-9E7D-B2143D59CAC4@freeswitch.org>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
	<A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
Message-ID: <20100315021746.GA31211@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Its nice to hear more people have this working.  The more that depend on
> this .. the simpler my job is... because If we break it they start to open
> jira's about it.

If both parties behind NAT could just go and register with
http://tunelbroker.net/ or http://www.freenet6.net/ or http://www.sixxs.net/
or a similar service, or lobby their ISPs to introduce native IPv6, this
problem would start to go away.




From spencer at 5ninesolutions.com  Sun Mar 14 19:20:50 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Sun, 14 Mar 2010 19:20:50 -0700
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
Message-ID: <3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>

So just in case anyone else ever needs to do this.. here's what I  
ended up with:

<include>
   <extension name="NNNNNNNNNNN-did">
     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="set" data="accountcode=<ACCTNUM>"/>
       <action application="bridge" data="sofia/default/$1${regex($ 
{sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}" />
     </condition>
   </extension>
</include>

Basically the extension UAs register to is created as the same as the  
account code.  We then use their ip at registration to dial a did on  
their end.

Spencer


On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:

> Duh.. :-)  Thanks!
>
> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>
>> Yes, sofia_contact with regex to extract the IP portion? You had  
>> the answer yourself.
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <spencer at 5ninesolutions.com 
>> > wrote:
>> I guess my question is, is there some function which will return the
>> IP address of a registered user?  Similar to sofia_contact but only
>> the IP address?
>>
>> Spencer
>>
>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>
>> > You sure can still have it call a registration.. You're mixing the
>> > dialplan and user directly as if they are a single entity.  They
>> > aren't.
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>> >
>> >> That looks promising, thanks.  I was hoping to have FS do it
>> >> automatically.  I.e. inbound dids are assigned to specific  
>> asterisk
>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>> >> inbound did @ whereever the registration came from instead of
>> >> having the IP address hard coded into a route.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Sun Mar 14 19:40:27 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 14 Mar 2010 23:40:27 -0300
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
	<3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
Message-ID: <ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>

Wiki is your friend to that type of info! Contribute to it, please!

JM


On Sun, Mar 14, 2010 at 11:20 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> So just in case anyone else ever needs to do this.. here's what I ended up
> with:
>
> <include>
>   <extension name="NNNNNNNNNNN-did">
>     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="set" data="accountcode=<ACCTNUM>"/>
>       <action application="bridge"
> data="sofia/default/$1${regex(${sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}"
> />
>     </condition>
>   </extension>
> </include>
>
> Basically the extension UAs register to is created as the same as the
> account code.  We then use their ip at registration to dial a did on their
> end.
>
> Spencer
>
>
> On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:
>
> Duh.. :-)  Thanks!
>
> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>
> Yes, sofia_contact with regex to extract the IP portion? You had the answer
> yourself.
>
> Jo?o Mesquita
>
>
>
> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I guess my question is, is there some function which will return the
>> IP address of a registered user?  Similar to sofia_contact but only
>> the IP address?
>>
>> Spencer
>>
>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>
>> > You sure can still have it call a registration.. You're mixing the
>> > dialplan and user directly as if they are a single entity.  They
>> > aren't.
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>> >
>> >> That looks promising, thanks.  I was hoping to have FS do it
>> >> automatically.  I.e. inbound dids are assigned to specific asterisk
>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>> >> inbound did @ whereever the registration came from instead of
>> >> having the IP address hard coded into a route.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Sun Mar 14 21:36:59 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 00:36:59 -0400
Subject: [Freeswitch-users] LUA and LUASQL Install
In-Reply-To: <4B9D4E2C.4020407@gmail.com>
References: <8a19bf2e1003141126i26065f9bnd45b870ad7475e52@mail.gmail.com>
	<8990C9A1-01CB-4907-BB1E-660EBD91EB28@5ninesolutions.com>
	<8a19bf2e1003141149j7fdfd07dw1639196fc22900d@mail.gmail.com>
	<8a19bf2e1003141246p2fcdf561kf5a8b22bdb2a2366@mail.gmail.com>
	<4B9D4E2C.4020407@gmail.com>
Message-ID: <8a19bf2e1003142136x17dbddd2j2c745b5265cd5a92@mail.gmail.com>

Thank you for the suggestion. Now FreeSWITCH , Lua , LuaSQL (ODBC)
working great.


On Sun, Mar 14, 2010 at 4:59 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  I would not use luasql with mysql driver because it leaks memory. Luasql
> odbc is better choice for production.
> What kind of errors do you have? If you installing luasql.odbc please make
> sure you have unix-odbc development packages.
>
> Nazim
>
>
>
>
>
> On 03/14/2010 11:46 PM, Aloysius Lloyd wrote:
>
> I am trying to install Luasql.  Lots of make errors.Here is the steps I
> follow
>
>  wget http://luaforge.net/frs/download.php/2686/luasql-2.1.1.tar.gz
> tar xfvz luasql-2.1.1.tar.gz
> cd luasql-2.1.1/
> vi config
>
>  uncomment the mysql
> enable mysql and change the mysql lib and include directory path.
>
>
>  How to install LuaSql in CentOS 5.4?
>
>  What is the best way to access mysql from Lua?
> odbc or natvie?
>
>  Thanks
> Lloyd
>
>
> On Sun, Mar 14, 2010 at 2:49 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Nazim & Spencer , thank you.
>>
>>  readline development package install solved the problem. Here is the Lua
>> install on CentOS 5.4
>>
>>  yum install readline-devel
>>  wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>> tar zxf lua-5.1.4.tar.gz
>> cd lua-5.1.4
>> make linux
>>  make linux install
>>
>>  Thanks
>> Lloyd
>>
>>
>>
>> On Sun, Mar 14, 2010 at 2:40 PM, Spencer Thomason <
>> spencer at 5ninesolutions.com> wrote:
>>
>>> I have Centos RPMS and source RPMS for lua 5.1.4 at
>>> http://repo.5ninesolutions.com
>>>   --
>>> *Spencer Thomason*
>>> 5Nine Solutions LLC
>>> e. sales at 5ninesolutions.com <spencer at 5ninesolutions.com>
>>> p. +1.888.271.7959
>>> f. +1.310.510.6980
>>>
>>>  <http://www.5ninesolutions.com/>
>>>
>>>   On Mar 14, 2010, at 11:26 AM, Aloysius Lloyd wrote:
>>>
>>>   Hi All,
>>>
>>>  Can some one help me to install Lua .
>>>
>>>  OS: CentOS 5.4
>>>
>>>
>>>  wget http://www.lua.org/ftp/lua-5.1.4.tar.gz
>>> tar zxf lua-5.1.4.tar.gz
>>> cd lua-5.1.4
>>> make linux
>>>
>>>  But getting error when running *make linux *
>>> *
>>> *
>>> *Here is the Output*
>>>
>>>  In file included from lua.h:16,
>>>                  from lua.c:15:
>>> luaconf.h:275:31: error: readline/readline.h: No such file or directory
>>> luaconf.h:276:30: error: readline/history.h: No such file or directory
>>> lua.c: In function ?pushline?:
>>> lua.c:182: warning: implicit declaration of function ?readline?
>>> lua.c:182: warning: assignment makes pointer from integer without a cast
>>> lua.c: In function ?loadline?:
>>> lua.c:210: warning: implicit declaration of function ?add_history?
>>> make[2]: *** [lua.o] Error 1
>>> make[2]: Leaving directory `/usr/src/lua-5.1.4/src'
>>> make[1]: *** [linux] Error 2
>>> make[1]: Leaving directory `/usr/src/lua-5.1.4/src'
>>> make: *** [linux] Error 2
>>>
>>>
>>>  Thank you in advance.
>>>
>>>
>>>  Thanks,
>>> Lloyd
>>>   _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tzury.by at reguluslabs.com  Sun Mar 14 23:52:39 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 08:52:39 +0200
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
Message-ID: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>

Hi,

Owe to an hardware failure I have had to reinstall my server.
Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap

Getting recommendation to try the new sangoma prid I tried to follow the
instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install

Being unable to make it work, I search through the mailing-list and saw the
specific versions which should be used.
Followed these instruction as well, yet, it is not working.

probing hardware shows all goos indications.

When I try to originate from within the FreeSWITCH's CLI I get

freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML public
2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
online.

-ERR NORMAL_CIRCUIT_CONGESTION

2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
freeswitch at snaptrunk-02>


when I try to run #smg_ctrl start
I get:

smg_ctrl: sangoma_prid failed to start
> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>

*However, that file ( /var/log/sangoma_mgd.log) is empty.

*I pastebin all relevant files at http://gist.github.com/332589*

*I am using the latest freeswitch (svn trunk)

I tried the smg with these versions (from
ftp://ftp.sangoma.com/linux/custom/DavidYS/)

   - wanpipe-3.5.10.smg_pri.1.tgz
   - wanpipe-3.5.10.3.smg_pri-v1.58.tgz

*

*
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From gmaruzz at celliax.org  Mon Mar 15 01:15:55 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Mar 2010 09:15:55 +0100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
Message-ID: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>

Ciao FreeSWITCHers,

is with pleasure that we're announcing the renaming of mod_skypiax
module to mod_skypopen.

The new wiki page is at
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk

To upgrade from a previous Skypiax installation, please read:
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Upgrading_from_Skypiax

Many many many thanks to the wonderful FreeSWITCH team that make it all happens!

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From jason at jasonjgw.net  Mon Mar 15 01:31:56 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 15 Mar 2010 19:31:56 +1100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
In-Reply-To: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>
References: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com>
Message-ID: <20100315083156.GA12318@jdc.jasonjgw.net>

Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Ciao FreeSWITCHers,
> 
> is with pleasure that we're announcing the renaming of mod_skypiax
> module to mod_skypopen.

Could you also rename it in the Debian packaging files (located in the debian
directory)? 




From gmaruzz at celliax.org  Mon Mar 15 01:47:56 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 15 Mar 2010 09:47:56 +0100
Subject: [Freeswitch-users] Skypiax has been renamed as Skypopen
In-Reply-To: <20100315083156.GA12318@jdc.jasonjgw.net>
References: <7b197bef1003150115la6f9540l77e0de601d1ce0f9@mail.gmail.com> 
	<20100315083156.GA12318@jdc.jasonjgw.net>
Message-ID: <7b197bef1003150147t7837e332jc2497197eaf2bc77@mail.gmail.com>

On Mon, Mar 15, 2010 at 9:31 AM, Jason White <jason at jasonjgw.net> wrote:
>
> Could you also rename it in the Debian packaging files (located in the debian
> directory)?

I'll ask the people who's taking care of Debian packages (I'll find
out who) to do it, thanks for reporting.

-giovanni

>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From maciej.aniserowicz at gmail.com  Mon Mar 15 01:56:36 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 15 Mar 2010 00:56:36 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL cause
Message-ID: <1268643396699-4735736.post@n2.nabble.com>


Hello,
I have the following problem:

I register with x-lite to my freeswitch and then call it by sending command
bgapi originate
{originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
&park()

Then i dial through a gateway which points to another freeswitch instance:
bgapi originate
{originate_timeout=20}[24cc_session_id=3134a5ae-ec00-4164-94b1-a7fb5e0368fa,ignore_early_media=true,max_forwards=999999,origination_uuid=a85fd47e-051f-42ea-987e-f4e090c7ae78]sofia/gateway/prospect_gateway/1300671139878747511456
&park()

The dialplan causes answering the phone, playing a file and hanging up.

The two calls are connected with this command:

bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline


Unfortunately after the second call hangs up, the first call gets hanged up
too.

I recently upgraded to FS trunk.

Logs are in pastebin:
first freeswitch: http://pastebin.freeswitch.org/12432
second freeswitch: http://pastebin.freeswitch.org/12433

Please take a look and help me figure this out..
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4735736.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From peter.olsson at visionutveckling.se  Mon Mar 15 02:20:03 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Mar 2010 10:20:03 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>

I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson

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From dujinfang at gmail.com  Mon Mar 15 03:08:30 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 15 Mar 2010 18:08:30 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
Message-ID: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>

ALL,

I just upgrade FS to 16992.

1) timer warning: How's possible the 10000 microseconds? audio sounds ok.


[WARNING] switch_time.c:206 Timer resolution of 10000 microseconds detected!
Do you have your kernel timer set to higher than 1 kHz? You may
experience audio problems.

uname -a
Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
2008 i686 GNU/Linux


I used a script to check the interupts

:~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
LOC:   14892854   14892843   14892841   14892830
LOC:   14892955   14892944   14892942   14892931
LOC:   14893056   14893045   14893043   14893032
LOC:   14893157   14893146   14893144   14893133
LOC:   14893258   14893247   14893245   14893234
LOC:   14893359   14893348   14893346   14893335
LOC:   14893460   14893449   14893447   14893436


2) run dingaling with client mode. dingaling doesn't pick up answer. I
originate a call from FS, the console blocks, gtalk client ring and it
looks ok when I hit answer, but no sound, FS console still blocks. a
few seconds later gtalk client shows:

Sorry! The voice chat with language_lesson failed because of a network
problem at 5:50 PM. Please try again.
 Click here to report this to Google.


here is a log with bgapi:
http://pastebin.freeswitch.org/12434


rev 14696 works on the same server same conf. another server running
16958M works at the same time. I have no clue to check, is it related
to the timer as in 1)? can some help me to take a look? let me know if
you need "dl_debug on" logs.

Thanks.



From dujinfang at gmail.com  Mon Mar 15 03:27:38 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 15 Mar 2010 18:27:38 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
Message-ID: <23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>

here is the client:

<profile type="client">
 <param name="name" value="gmail.com"/>
 <param name="login" value="xx at gmail.com/talk"/>
 <param name="password" value="password"/>
 <param name="server" value="talk.google.com"/>
 <param name="dialplan" value="XML"/>
 <param name="contex" value="public"/>
 <param name="message" value="FreeSWITCH"/>
 <param name="rtp-ip" value="auto"/>
 <!-- if you are behind NAT, you should probably use stun -->
 <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
 <param name="auto-login" value="true"/>
 <param name="auto-reply" value="Welcome"/>
 <param name="sasl" value="plain"/>
 <param name="tls" value="true"/>
 <param name="exten" value="5000"/>
<!-- <param name="local-network-acl" value="localnet.auto"/> -->
</profile>

2010/3/15 Seven Du <dujinfang at gmail.com>:
> ALL,
>
> I just upgrade FS to 16992.
>
> 1) timer warning: How's possible the 10000 microseconds? audio sounds ok.
>
>
> [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds detected!
> Do you have your kernel timer set to higher than 1 kHz? You may
> experience audio problems.
>
> uname -a
> Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> 2008 i686 GNU/Linux
>
>
> I used a script to check the interupts
>
> :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
> LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
> LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
> LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
> LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
> LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
> LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>
>
> 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> originate a call from FS, the console blocks, gtalk client ring and it
> looks ok when I hit answer, but no sound, FS console still blocks. a
> few seconds later gtalk client shows:
>
> Sorry! The voice chat with language_lesson failed because of a network
> problem at 5:50 PM. Please try again.
> ?Click here to report this to Google.
>
>
> here is a log with bgapi:
> http://pastebin.freeswitch.org/12434
>
>
> rev 14696 works on the same server same conf. another server running
> 16958M works at the same time. I have no clue to check, is it related
> to the timer as in 1)? can some help me to take a look? let me know if
> you need "dl_debug on" logs.
>
> Thanks.
>



From tzury.by at reguluslabs.com  Mon Mar 15 03:28:08 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 12:28:08 +0200
Subject: [Freeswitch-users] regarding the recent sangoma boost
Message-ID: <10128ef11003150328t3e033993nb4de567d9e129bd6@mail.gmail.com>

one must run # mkdir /root/sangoma before head.
otherwise, sangoma_prid will fail to start


I discoverd this since /var/log/sangoma_mgd.log file was empty even though
the line
local2.*    /var/log/sangoma_mgd.log
exists in /etc/syslog.conf

I did the following:
# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# /usr/sbin/sangoma_prid &> sangoma_prid.log

an then with # tail -f sangoma_prid.log
I got
==> sangoma_prid.log <==
================System restart=============
= Sangoma PRI Protocol Stack Daemon       =
= Version: 1.58                           =
= Date: Feb 18 2010                       =
= Wanpipe Release: wanpipe-3.5.10.3          =
= Revision:Revision: 15364          =
===========================================
sh: gen_fs_gcore.sh: not found
touch: cannot touch `/root/sangoma/sangoma_prid.gcore.24821': No such file
or directory
/usr/bin/gcore: 83: cannot create /root/sangoma/sangoma_prid.gcore.24821:
Directory nonexistent
gcore: failed to create /root/sangoma/sangoma_prid.gcore.24794


I then ran `# mkdir /root/sangoma` and restart everything

Right now I am facing another problem which hope to over come right after
lunch

the problem is this error

freeswitch at snaptrunk-02> originate openzap/1/a/0525133399 XML public
2010-03-15 12:20:56.307066 [WARNING] ozmod_sangoma_boost.c:341 TX EVENT:
CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[N/A] Cd=[0525133399] Ci=[N/A]
Rdnis=[]



2010-03-15 12:21:06.858887 [WARNING] sangoma_boost_client.c:221 TX EVENT
(N): CALL_START_NACK:(82) [w1g1] Rc=0 CSid=1 Seq=1
2010-03-15 12:21:06.858887 [ERR] switch_ivr_originate.c:2422 Cannot create
outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE]

-ERR RECOVERY_ON_TIMER_EXPIRE

2010-03-15 12:21:06.858887 [DEBUG] switch_ivr_originate.c:3220 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
freeswitch at snaptrunk-02>
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From jbrucehopkins at gmail.com  Mon Mar 15 04:39:26 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Mon, 15 Mar 2010 11:39:26 +0000
Subject: [Freeswitch-users] doublenat outgoing call issue
In-Reply-To: <20100315021746.GA31211@jdc.jasonjgw.net>
References: <d03a10e01003080703p42e7b71cwfdbbedf7a442d72e@mail.gmail.com>
	<d03a10e01003080912h6856efdqb6344b8d1724b12e@mail.gmail.com>
	<8215DEFE-460E-4764-BF96-35FBE820EC11@freeswitch.org>
	<63837C30-11FB-47E6-8173-3AFEA76A5BB4@gmail.com>
	<000001cabfbb$f2114c90$d633e5b0$@co.za>
	<d03a10e01003131456l68ca62e6wd7c2989cdb6f8975@mail.gmail.com>
	<028601cac3a6$056fc940$104f5bc0$@co.za>
	<d03a10e01003141853l30255ef6jd1aa486f77f60cc1@mail.gmail.com>
	<A001FD5F-16BB-4C13-B021-B10CC43D4B33@freeswitch.org>
	<20100315021746.GA31211@jdc.jasonjgw.net>
Message-ID: <d03a10e01003150439q334b88d9t5f18f97bc6e4fcee@mail.gmail.com>

Very interesting.  Thanks for sharing this.  I must read up on it.

cheersBruce

On 15 March 2010 02:17, Jason White <jason at jasonjgw.net> wrote:

> Brian West <brian at freeswitch.org> wrote:
> > Its nice to hear more people have this working.  The more that depend on
> > this .. the simpler my job is... because If we break it they start to
> open
> > jira's about it.
>
> If both parties behind NAT could just go and register with
> http://tunelbroker.net/ or http://www.freenet6.net/ or
> http://www.sixxs.net/
> or a similar service, or lobby their ISPs to introduce native IPv6, this
> problem would start to go away.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lakindia89 at gmail.com  Mon Mar 15 05:38:56 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Mon, 15 Mar 2010 18:08:56 +0530
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
In-Reply-To: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
References: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
Message-ID: <7d79b3931003150538u4621c056l2469f77fd78d0ad7@mail.gmail.com>

Make sure that the following packages are installed.

apt-get install libsctp1 patch bison build-essential gcc g++ make
libncurses5-dev flex linux-headers-$(uname -r) automake libtools autoconf
libtermcap-devel devscripts pbuilder glibc2.4



On Mon, Mar 15, 2010 at 12:22 PM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

> Hi,
>
> Owe to an hardware failure I have had to reinstall my server.
> Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap
>
> Getting recommendation to try the new sangoma prid I tried to follow the
> instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> Being unable to make it work, I search through the mailing-list and saw the
> specific versions which should be used.
> Followed these instruction as well, yet, it is not working.
>
> probing hardware shows all goos indications.
>
> When I try to originate from within the FreeSWITCH's CLI I get
>
> freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML
> public
> 2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
> online.
>
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> freeswitch at snaptrunk-02>
>
>
> when I try to run #smg_ctrl start
> I get:
>
> smg_ctrl: sangoma_prid failed to start
>> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>>
>
> *However, that file ( /var/log/sangoma_mgd.log) is empty.
>
> *I pastebin all relevant files at http://gist.github.com/332589*
>
> *I am using the latest freeswitch (svn trunk)
>
> I tried the smg with these versions (from
> ftp://ftp.sangoma.com/linux/custom/DavidYS/)
>
>    - wanpipe-3.5.10.smg_pri.1.tgz
>    - wanpipe-3.5.10.3.smg_pri-v1.58.tgz
>
> *
>
> *
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Mon Mar 15 06:40:25 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 09:40:25 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
Message-ID: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>

Hi All,

I have the following simple Lua script answer a inbound call then dial
another number and bridge the call.

Is there any way to get the  call duration or billing sec using the

session:getVariable(  );



-- answer the call
session:answer();

-- sleep a second
session:sleep(1000);

-- Initiate an outbound call
obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)

obSession:execute("sched_hangup","+60 alloted_timeout");

-- bride the call
    freeswitch.bridge(session, obSession);

-- hangup
session:hangup();


Thank you
Lloyd
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From tzury.by at reguluslabs.com  Mon Mar 15 06:45:51 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 15:45:51 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error using
	Sangoma (sangoma_prid)
Message-ID: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>

Hi,

This is my setup

FreeSWITCH: Version 1.0.trunk (16992M)
Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58

I have finally managed to set up the environment coping many hick-ups
reported in this mailing list recently.
However, I am still encountering problems when trying to perform call from
SIP to PRI.

I have configured my dialplan to bridge all numbers which are not

pastebin for all configuration files is available at:
http://pastebin.freeswitch.org/12437

pastebin for CLI output is available at:
http://pastebin.freeswitch.org/12436
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From tzury.by at reguluslabs.com  Mon Mar 15 06:51:51 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 15:51:51 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
Message-ID: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>

Hi,

This is my setup

FreeSWITCH: Version 1.0.trunk (16992M)
Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58

I have finally managed to set up the environment coping many hick-ups
reported in this mailing list recently.
However, I am still encountering problems when trying to perform call
from SIP to PRI.

I have configured my dialplan to bridge all numbers which are not SIP
to be bridged over to the PSTN.
However, when calling a PSTN number I am getting
RECOVERY_ON_TIMER_EXPIRE error

pastebin for all configuration files is available at:
http://pastebin.freeswitch.org/12437

pastebin for CLI output is available at:
http://pastebin.freeswitch.org/12436



From sos at sokhapkin.dyndns.org  Mon Mar 15 06:56:37 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 09:56:37 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
Message-ID: <201003150956.37356.sos@sokhapkin.dyndns.org>

Unfortunately all "interesting" billing-related channel variables are set 
after the lua script execution, when channel enters REPORTING state. To access 
those variables I execute separate lua script using mod_cdr_csv, it allows to 
invoke lua script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> Hi All,
> 
> I have the following simple Lua script answer a inbound call then dial
> another number and bridge the call.
> 
> Is there any way to get the  call duration or billing sec using the
> 
> session:getVariable(  );
> 
> 
> 
> -- answer the call
> session:answer();
> 
> -- sleep a second
> session:sleep(1000);
> 
> -- Initiate an outbound call
> obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> 
> obSession:execute("sched_hangup","+60 alloted_timeout");
> 
> -- bride the call
>     freeswitch.bridge(session, obSession);
> 
> -- hangup
> session:hangup();
> 
> 
> Thank you
> Lloyd
> 




From moises.silva at gmail.com  Mon Mar 15 07:00:20 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 10:00:20 -0400
Subject: [Freeswitch-users] errors and failures at snagoma+freeswitch
In-Reply-To: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
References: <10128ef11003142352h7b151bc6me0f958f34fc00071@mail.gmail.com>
Message-ID: <c4d05cbe1003150700k455a1a5ag82668bee64d19f5c@mail.gmail.com>

Enable debug and do svn update. I suspect you don't have sctp.

You can also run sangoma_prid directly to see the output in stdout/stderr

# export LD_LIBRARY_PATH=/usr/lib/sangoma_prid
# sangoma_prid

If there is no output in /var/log may be is because /etc/syslog.conf is
missing something like this:

# Sangoma Media Gateway log
local2.*                /var/log/sangoma_mgd.log


# Sangoma BRI Daemon (smg_bri)  log
local3.*                /var/log/sangoma_bri.log


On Mon, Mar 15, 2010 at 2:52 AM, Tzury Bar Yochay
<tzury.by at reguluslabs.com>wrote:

> Hi,
>
> Owe to an hardware failure I have had to reinstall my server.
> Originally I have had the combination of FreeSWITCH/Sangoma/LibPRI/OpenZap
>
> Getting recommendation to try the new sangoma prid I tried to follow the
> instructions found at http://wiki.sangoma.com/wanpipe-freeswitch-install
>
> Being unable to make it work, I search through the mailing-list and saw the
> specific versions which should be used.
> Followed these instruction as well, yet, it is not working.
>
> probing hardware shows all goos indications.
>
> When I try to originate from within the FreeSWITCH's CLI I get
>
> freeswitch at snaptrunk-02> originate openzap/smg_prid/a/0525133399 XML
> public
> 2010-03-14 16:09:14.575286 [CRIT] ozmod_sangoma_boost.c:254 SPAN is not
> online.
>
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-03-14 16:09:14.575286 [ERR] switch_ivr_originate.c:2422 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
> freeswitch at snaptrunk-02>
>
>
> when I try to run #smg_ctrl start
> I get:
>
> smg_ctrl: sangoma_prid failed to start
>> smg_ctrl: check /var/log/sangoma_mgd.log for errors
>>
>
> *However, that file ( /var/log/sangoma_mgd.log) is empty.
>
> *I pastebin all relevant files at http://gist.github.com/332589*
>
> *I am using the latest freeswitch (svn trunk)
>
> I tried the smg with these versions (from
> ftp://ftp.sangoma.com/linux/custom/DavidYS/)
>
>    - wanpipe-3.5.10.smg_pri.1.tgz
>    - wanpipe-3.5.10.3.smg_pri-v1.58.tgz
>
> *
>
> *
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Mon Mar 15 07:03:22 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 10:03:22 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
Message-ID: <c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>

take an isdn pcap dump of the wanpipe interface to see the messages there (I
think you already know how, if not, search "pcap" in wiki.sangoma.com)

On Mon, Mar 15, 2010 at 9:51 AM, Tzury Bar Yochay
<tzury.by at reguluslabs.com>wrote:

> Hi,
>
> This is my setup
>
> FreeSWITCH: Version 1.0.trunk (16992M)
> Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58
>
> I have finally managed to set up the environment coping many hick-ups
> reported in this mailing list recently.
> However, I am still encountering problems when trying to perform call
> from SIP to PRI.
>
> I have configured my dialplan to bridge all numbers which are not SIP
> to be bridged over to the PSTN.
> However, when calling a PSTN number I am getting
> RECOVERY_ON_TIMER_EXPIRE error
>
> pastebin for all configuration files is available at:
> http://pastebin.freeswitch.org/12437
>
> pastebin for CLI output is available at:
> http://pastebin.freeswitch.org/12436
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From lukasz at voiceworks.pl  Mon Mar 15 07:11:51 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 15 Mar 2010 15:11:51 +0100
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
 using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
Message-ID: <4B9E4027.4000100@voiceworks.pl>

I would check wanpipe 3.5.10.9 first. I think I had similar problem when 
I installed new openzap and older sangoma_prid (older wanpipe).

Tzury Bar Yochay wrote:
> Hi,
>
> This is my setup
>
> FreeSWITCH: Version 1.0.trunk (16992M)
> Sangoma: wanpipe-3.5.10.3.smg_pri-v1.58
>
> I have finally managed to set up the environment coping many hick-ups
> reported in this mailing list recently.
> However, I am still encountering problems when trying to perform call
> from SIP to PRI.
>
> I have configured my dialplan to bridge all numbers which are not SIP
> to be bridged over to the PSTN.
> However, when calling a PSTN number I am getting
> RECOVERY_ON_TIMER_EXPIRE error
>
> pastebin for all configuration files is available at:
> http://pastebin.freeswitch.org/12437
>
> pastebin for CLI output is available at:
> http://pastebin.freeswitch.org/12436
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From tzury.by at reguluslabs.com  Mon Mar 15 07:11:58 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 16:11:58 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<c4d05cbe1003150703g19a5f8a0u60fc9b7332fd439@mail.gmail.com>
Message-ID: <10128ef11003150711q571f28fescf478066e3b3b209@mail.gmail.com>

On Mon, Mar 15, 2010 at 4:03 PM, Moises Silva <moises.silva at gmail.com> wrote:
> take an isdn pcap dump of the wanpipe interface to see the messages there (I
> think you already know how, if not, search "pcap" in wiki.sangoma.com)
pcap file attached, also available at
http://rapidshare.com/files/363660904/isdn.pcap.html
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From tzury.by at reguluslabs.com  Mon Mar 15 07:21:44 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 15 Mar 2010 16:21:44 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <4B9E4027.4000100@voiceworks.pl>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<4B9E4027.4000100@voiceworks.pl>
Message-ID: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>

On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl> wrote:
> I would check wanpipe 3.5.10.9 first. I think I had similar problem when
> I installed new openzap and older sangoma_prid (older wanpipe).

Bu then I might face this issue
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
which was the reason I went to this version at first place



From lloyd.aloysius at gmail.com  Mon Mar 15 07:34:22 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 10:34:22 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003150956.37356.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>

How to execute another lua script when channel enter the REPORTING state?

What are the channel variable names?

Thanks
Lloyd

On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Unfortunately all "interesting" billing-related channel variables are set
> after the lua script execution, when channel enters REPORTING state. To
> access
> those variables I execute separate lua script using mod_cdr_csv, it allows
> to
> invoke lua script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > Hi All,
> >
> > I have the following simple Lua script answer a inbound call then dial
> > another number and bridge the call.
> >
> > Is there any way to get the  call duration or billing sec using the
> >
> > session:getVariable(  );
> >
> >
> >
> > -- answer the call
> > session:answer();
> >
> > -- sleep a second
> > session:sleep(1000);
> >
> > -- Initiate an outbound call
> > obSession =
> freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> >
> > obSession:execute("sched_hangup","+60 alloted_timeout");
> >
> > -- bride the call
> >     freeswitch.bridge(session, obSession);
> >
> > -- hangup
> > session:hangup();
> >
> >
> > Thank you
> > Lloyd
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lukasz at voiceworks.pl  Mon Mar 15 07:36:47 2010
From: lukasz at voiceworks.pl (Lukasz Kutkowski)
Date: Mon, 15 Mar 2010 15:36:47 +0100
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
 using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
Message-ID: <4B9E45FF.3030204@voiceworks.pl>

I'm running

FreeSWITCH Version 1.0.trunk (16966)
WANPIPE Release: 3.5.10.9

in my lab and everything is ok. Check 
http://wiki.sangoma.com/wanpipe-freeswitch-install. They recommend 
wanpipe 3.5.10.9. I think you should try it.

Tzury Bar Yochay wrote:
> On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl> wrote:
>   
>> I would check wanpipe 3.5.10.9 first. I think I had similar problem when
>> I installed new openzap and older sangoma_prid (older wanpipe).
>>     
>
> Bu then I might face this issue
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
> which was the reason I went to this version at first place
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From sos at sokhapkin.dyndns.org  Mon Mar 15 07:48:34 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 10:48:34 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
Message-ID: <201003151048.34689.sos@sokhapkin.dyndns.org>

cdr_csv.conf.xml:

<param name="default-template" value="lua"/>
<templates>
  <template name="lua">${lua(onhangup.lua)}</template>

All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv are 
accessible from that lua script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> How to execute another lua script when channel enter the REPORTING state?
> 
> What are the channel variable names?
> 
> Thanks
> Lloyd
> 
> On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Unfortunately all "interesting" billing-related channel variables are set
> > after the lua script execution, when channel enters REPORTING state. To
> > access
> > those variables I execute separate lua script using mod_cdr_csv, it
> > allows to
> > invoke lua script.
> >
> > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > Hi All,
> > >
> > > I have the following simple Lua script answer a inbound call then dial
> > > another number and bridge the call.
> > >
> > > Is there any way to get the  call duration or billing sec using the
> > >
> > > session:getVariable(  );
> > >
> > >
> > >
> > > -- answer the call
> > > session:answer();
> > >
> > > -- sleep a second
> > > session:sleep(1000);
> > >
> > > -- Initiate an outbound call
> > > obSession =
> >
> > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> >
> > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > >
> > > -- bride the call
> > >     freeswitch.bridge(session, obSession);
> > >
> > > -- hangup
> > > session:hangup();
> > >
> > >
> > > Thank you
> > > Lloyd
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 




From ahmed.ajmal at breezecom.ae  Mon Mar 15 07:50:49 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Mon, 15 Mar 2010 19:50:49 +0500
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
References: <016f01cac05a$36be9610$a43bc230$@ajmal@breezecom.ae>
	<D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
Message-ID: <016201cac44e$eadf33f0$c09d9bd0$@ajmal@breezecom.ae>

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Thanks

Ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, March 10, 2010 7:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

We already set you a variable called sip_gateway_name when a call is placed
out a gateway.  You're also doing this wrong you should NEVER EVER do
@$ipgwX as that data is in the gateway.  You should simply do
sofia/gateway/gwname/number

 

/b

 

On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:





Hi

 

I am trying to set a channel variable on the bridge application using this:

 

<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>

 

The application runs fine, what I now need to do is set/export the 'gw'
channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.

 

Thanks

Ahmed

 

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From lloyd.aloysius at gmail.com  Mon Mar 15 07:54:13 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 10:54:13 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003151048.34689.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003150956.37356.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150734q41a442dcl40261b59ff46c58b@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>

How to trigger this template from the main script ? or Does this
automatically trigger when finish the first script complete?

Thanks
Lloyd

On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> cdr_csv.conf.xml:
>
> <param name="default-template" value="lua"/>
> <templates>
>  <template name="lua">${lua(onhangup.lua)}</template>
>
> All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv are
> accessible from that lua script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > How to execute another lua script when channel enter the REPORTING state?
> >
> > What are the channel variable names?
> >
> > Thanks
> > Lloyd
> >
> > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > Unfortunately all "interesting" billing-related channel variables are
> set
> > > after the lua script execution, when channel enters REPORTING state. To
> > > access
> > > those variables I execute separate lua script using mod_cdr_csv, it
> > > allows to
> > > invoke lua script.
> > >
> > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > Hi All,
> > > >
> > > > I have the following simple Lua script answer a inbound call then
> dial
> > > > another number and bridge the call.
> > > >
> > > > Is there any way to get the  call duration or billing sec using the
> > > >
> > > > session:getVariable(  );
> > > >
> > > >
> > > >
> > > > -- answer the call
> > > > session:answer();
> > > >
> > > > -- sleep a second
> > > > session:sleep(1000);
> > > >
> > > > -- Initiate an outbound call
> > > > obSession =
> > >
> > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > >
> > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > >
> > > > -- bride the call
> > > >     freeswitch.bridge(session, obSession);
> > > >
> > > > -- hangup
> > > > session:hangup();
> > > >
> > > >
> > > > Thank you
> > > > Lloyd
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
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From sos at sokhapkin.dyndns.org  Mon Mar 15 08:02:03 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 15 Mar 2010 11:02:03 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
Message-ID: <201003151102.03407.sos@sokhapkin.dyndns.org>

That script will be executed automatically when call ends. You can set some 
channel variables in the main lua script and access those variables from 
hangup lua script if you need to pass the information to the hangup script.

On Monday 15 March 2010, Aloysius Lloyd wrote:
> How to trigger this template from the main script ? or Does this
> automatically trigger when finish the first script complete?
> 
> Thanks
> Lloyd
> 
> On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > cdr_csv.conf.xml:
> >
> > <param name="default-template" value="lua"/>
> > <templates>
> >  <template name="lua">${lua(onhangup.lua)}</template>
> >
> > All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv
> > are accessible from that lua script.
> >
> > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > How to execute another lua script when channel enter the REPORTING
> > > state?
> > >
> > > What are the channel variable names?
> > >
> > > Thanks
> > > Lloyd
> > >
> > > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> > >
> > > <sos at sokhapkin.dyndns.org>wrote:
> > > > Unfortunately all "interesting" billing-related channel variables are
> >
> > set
> >
> > > > after the lua script execution, when channel enters REPORTING state.
> > > > To access
> > > > those variables I execute separate lua script using mod_cdr_csv, it
> > > > allows to
> > > > invoke lua script.
> > > >
> > > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > > Hi All,
> > > > >
> > > > > I have the following simple Lua script answer a inbound call then
> >
> > dial
> >
> > > > > another number and bridge the call.
> > > > >
> > > > > Is there any way to get the  call duration or billing sec using the
> > > > >
> > > > > session:getVariable(  );
> > > > >
> > > > >
> > > > >
> > > > > -- answer the call
> > > > > session:answer();
> > > > >
> > > > > -- sleep a second
> > > > > session:sleep(1000);
> > > > >
> > > > > -- Initiate an outbound call
> > > > > obSession =
> > > >
> > > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > > >
> > > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > > >
> > > > > -- bride the call
> > > > >     freeswitch.bridge(session, obSession);
> > > > >
> > > > > -- hangup
> > > > > session:hangup();
> > > > >
> > > > >
> > > > > Thank you
> > > > > Lloyd
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> 




From moises.silva at gmail.com  Mon Mar 15 08:28:13 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 15 Mar 2010 11:28:13 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
Message-ID: <c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>

Ok, so the call is not even being placed in the ISDN network.

Did you manage to get the sangoma_prid logs working? if that's the case then
paste them too, it seems sangoma_prid is not seeing the request from
FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
response from sangoma_prid binary (they both communicate through sctp
socket).
-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Mon, Mar 15, 2010 at 10:21 AM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

> On Mon, Mar 15, 2010 at 4:11 PM, Lukasz Kutkowski <lukasz at voiceworks.pl>
> wrote:
> > I would check wanpipe 3.5.10.9 first. I think I had similar problem when
> > I installed new openzap and older sangoma_prid (older wanpipe).
>
> Bu then I might face this issue
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-February/025361.html
> which was the reason I went to this version at first place
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Mar 15 08:31:03 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 09:31:03 -0600
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
Message-ID: <191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>

you could wait for the execute_complete event for your break command.


On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  I?m not sure if this belongs in Jira, or if it?s possible to do it
> another way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From srinivas.ksvreddy at gmail.com  Mon Mar 15 08:36:29 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Mon, 15 Mar 2010 11:36:29 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
Message-ID: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>

Hi,

i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
like just farwading the call.
<action application="bridge" data=
sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
/ <data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>>
i do not using any gateway here, just farwarding call to router, is this
correct way of doing? any idea

-- 
Srinivasula Reddy K
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From peter.olsson at visionutveckling.se  Mon Mar 15 08:40:48 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Mar 2010 16:40:48 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>

Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.

On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
!DSPAM:4b9e543132932880416179!
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From anthony.minessale at gmail.com  Mon Mar 15 08:45:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 09:45:37 -0600
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
Message-ID: <191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>

That seems suspiciously similar to an old issue we have already fixed, how
exactly did you rebuild?


On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I rebuilt today.
>
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>
> > Are you sure you're on SVN trunk?
> >
> > /b
> >
> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
> >
> >> No.  Its on our hardware, we just use xen to virtualize the
> >> machines.. this is the only vm on the machine.  Gcore complains
> >> about no debugging symbols found, do I need to rebuild?
> >>
> >> It appears to be a problem when registering to one of my trunks.
> >> The very strange thing is the everything will be working perfectly
> >> and then all of a sudden this happens, i.e. trunks regged, calls
> >> completing, etc. This is a very lightly loaded server.
> >>
> >> This is the output from the console:
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Mon Mar 15 09:21:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:21:15 -0600
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
Message-ID: <191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>

you could also wait for the execute_complete of the file you were playing

On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
>  On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> !DSPAM:4b9e543132932880416179!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Mar 15 09:28:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:28:05 -0600
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
Message-ID: <191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>

you would have to execute

dl_debug on

to get a trace that would tell us anything about that?

Are you using the windows gtalk client or the web site?
We are following google who have been pushing for the online client embedded
in the gmail website.




On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:

> here is the client:
>
> <profile type="client">
>  <param name="name" value="gmail.com"/>
>  <param name="login" value="xx at gmail.com/talk"/>
>  <param name="password" value="password"/>
>  <param name="server" value="talk.google.com"/>
>  <param name="dialplan" value="XML"/>
>  <param name="contex" value="public"/>
>  <param name="message" value="FreeSWITCH"/>
>  <param name="rtp-ip" value="auto"/>
>  <!-- if you are behind NAT, you should probably use stun -->
>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>  <param name="auto-login" value="true"/>
>  <param name="auto-reply" value="Welcome"/>
>  <param name="sasl" value="plain"/>
>  <param name="tls" value="true"/>
>  <param name="exten" value="5000"/>
> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> </profile>
>
> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> > ALL,
> >
> > I just upgrade FS to 16992.
> >
> > 1) timer warning: How's possible the 10000 microseconds? audio sounds ok.
> >
> >
> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> detected!
> > Do you have your kernel timer set to higher than 1 kHz? You may
> > experience audio problems.
> >
> > uname -a
> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> > 2008 i686 GNU/Linux
> >
> >
> > I used a script to check the interupts
> >
> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> > LOC:   14892854   14892843   14892841   14892830
> > LOC:   14892955   14892944   14892942   14892931
> > LOC:   14893056   14893045   14893043   14893032
> > LOC:   14893157   14893146   14893144   14893133
> > LOC:   14893258   14893247   14893245   14893234
> > LOC:   14893359   14893348   14893346   14893335
> > LOC:   14893460   14893449   14893447   14893436
> >
> >
> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> > originate a call from FS, the console blocks, gtalk client ring and it
> > looks ok when I hit answer, but no sound, FS console still blocks. a
> > few seconds later gtalk client shows:
> >
> > Sorry! The voice chat with language_lesson failed because of a network
> > problem at 5:50 PM. Please try again.
> >  Click here to report this to Google.
> >
> >
> > here is a log with bgapi:
> > http://pastebin.freeswitch.org/12434
> >
> >
> > rev 14696 works on the same server same conf. another server running
> > 16958M works at the same time. I have no clue to check, is it related
> > to the timer as in 1)? can some help me to take a look? let me know if
> > you need "dl_debug on" logs.
> >
> > Thanks.
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Mon Mar 15 09:39:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 10:39:26 -0600
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268643396699-4735736.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
Message-ID: <191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>

according to the code:

you are bridging 2 channels where one of them is not answered yet.
and if it never is answered before the timeout it will hangup.

try waiting for both to be answered before you bridge them.



On Mon, Mar 15, 2010 at 2:56 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello,
> I have the following problem:
>
> I register with x-lite to my freeswitch and then call it by sending command
> bgapi originate
>
> {originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
> &park()
>
> Then i dial through a gateway which points to another freeswitch instance:
> bgapi originate
>
> {originate_timeout=20}[24cc_session_id=3134a5ae-ec00-4164-94b1-a7fb5e0368fa,ignore_early_media=true,max_forwards=999999,origination_uuid=a85fd47e-051f-42ea-987e-f4e090c7ae78]sofia/gateway/prospect_gateway/1300671139878747511456
> &park()
>
> The dialplan causes answering the phone, playing a file and hanging up.
>
> The two calls are connected with this command:
>
> bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
> intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline
>
>
> Unfortunately after the second call hangs up, the first call gets hanged up
> too.
>
> I recently upgraded to FS trunk.
>
> Logs are in pastebin:
> first freeswitch: http://pastebin.freeswitch.org/12432
> second freeswitch: http://pastebin.freeswitch.org/12433
>
> Please take a look and help me figure this out..
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4735736.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From dujinfang at gmail.com  Mon Mar 15 10:01:08 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 01:01:08 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
Message-ID: <23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>

Thanks, It's late now, I will get logs tomorrow. I tested both on
Windows client and web site on Mac. They both work on the other two
instances of FreeSWITCH I mentioned in this thread.

btw, can you give me a hint on the timer warning?

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> you would have to execute
>
> dl_debug on
>
> to get a trace that would tell us anything about that?
>
> Are you using the windows gtalk client or the web site?
> We are following google who have been pushing for the online client embedded
> in the gmail website.
>
>
>
>
> On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> here is the client:
>>
>> <profile type="client">
>> ?<param name="name" value="gmail.com"/>
>> ?<param name="login" value="xx at gmail.com/talk"/>
>> ?<param name="password" value="password"/>
>> ?<param name="server" value="talk.google.com"/>
>> ?<param name="dialplan" value="XML"/>
>> ?<param name="contex" value="public"/>
>> ?<param name="message" value="FreeSWITCH"/>
>> ?<param name="rtp-ip" value="auto"/>
>> ?<!-- if you are behind NAT, you should probably use stun -->
>> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> ?<param name="auto-login" value="true"/>
>> ?<param name="auto-reply" value="Welcome"/>
>> ?<param name="sasl" value="plain"/>
>> ?<param name="tls" value="true"/>
>> ?<param name="exten" value="5000"/>
>> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> </profile>
>>
>> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> > ALL,
>> >
>> > I just upgrade FS to 16992.
>> >
>> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
>> > ok.
>> >
>> >
>> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> > detected!
>> > Do you have your kernel timer set to higher than 1 kHz? You may
>> > experience audio problems.
>> >
>> > uname -a
>> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
>> > 2008 i686 GNU/Linux
>> >
>> >
>> > I used a script to check the interupts
>> >
>> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >
>> >
>> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
>> > originate a call from FS, the console blocks, gtalk client ring and it
>> > looks ok when I hit answer, but no sound, FS console still blocks. a
>> > few seconds later gtalk client shows:
>> >
>> > Sorry! The voice chat with language_lesson failed because of a network
>> > problem at 5:50 PM. Please try again.
>> > ?Click here to report this to Google.
>> >
>> >
>> > here is a log with bgapi:
>> > http://pastebin.freeswitch.org/12434
>> >
>> >
>> > rev 14696 works on the same server same conf. another server running
>> > 16958M works at the same time. I have no clue to check, is it related
>> > to the timer as in 1)? can some help me to take a look? let me know if
>> > you need "dl_debug on" logs.
>> >
>> > Thanks.
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From maciej.aniserowicz at gmail.com  Mon Mar 15 10:04:04 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 15 Mar 2010 09:04:04 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
Message-ID: <1268672644810-4738333.post@n2.nabble.com>


Hi Anthony,
Sorry if this was unclear - I am actually waiting for the phone to be
answered. The two calls are bridged in handler for ANSWERED event of the 2nd
call...
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4738333.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Mon Mar 15 10:04:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 09:04:46 -0800
Subject: [Freeswitch-users] not answering calls on TDM400 port
In-Reply-To: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>
References: <3c5cf5261003131753y454a419bp609ed55342f8ddb0@mail.gmail.com>
Message-ID: <87f2f3b91003151004j4c91f78cie800f9ce8be2eef4@mail.gmail.com>

The TDM400 is a zaptel/dahdi card so you can use zttool (or the dahdi
equivalent) to see what the channels are doing.
-MC

On Sat, Mar 13, 2010 at 5:53 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> I noticed Freeswitch wasn't answering calls on one of the TDM400 ports
> yesterday.
>
> All other ports seemed to be OK.
>
> Nothing showed up on the console, set at debug log level.
>
> I restarted Freeswitch and it came good.
>
> What should I do if it happens again? Is there any debugging
> information I can extract from the system? If so how?
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Mar 15 10:18:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 11:18:06 -0600
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
Message-ID: <191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>

FS would prefer you run your kernel at 1000hz you are probably on a VM if
its 10000hz


On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:

> Thanks, It's late now, I will get logs tomorrow. I tested both on
> Windows client and web site on Mac. They both work on the other two
> instances of FreeSWITCH I mentioned in this thread.
>
> btw, can you give me a hint on the timer warning?
>
> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > you would have to execute
> >
> > dl_debug on
> >
> > to get a trace that would tell us anything about that?
> >
> > Are you using the windows gtalk client or the web site?
> > We are following google who have been pushing for the online client
> embedded
> > in the gmail website.
> >
> >
> >
> >
> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> here is the client:
> >>
> >> <profile type="client">
> >>  <param name="name" value="gmail.com"/>
> >>  <param name="login" value="xx at gmail.com/talk"/>
> >>  <param name="password" value="password"/>
> >>  <param name="server" value="talk.google.com"/>
> >>  <param name="dialplan" value="XML"/>
> >>  <param name="contex" value="public"/>
> >>  <param name="message" value="FreeSWITCH"/>
> >>  <param name="rtp-ip" value="auto"/>
> >>  <!-- if you are behind NAT, you should probably use stun -->
> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >>  <param name="auto-login" value="true"/>
> >>  <param name="auto-reply" value="Welcome"/>
> >>  <param name="sasl" value="plain"/>
> >>  <param name="tls" value="true"/>
> >>  <param name="exten" value="5000"/>
> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> </profile>
> >>
> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> > ALL,
> >> >
> >> > I just upgrade FS to 16992.
> >> >
> >> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
> >> > ok.
> >> >
> >> >
> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> >> > detected!
> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> > experience audio problems.
> >> >
> >> > uname -a
> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
> >> > 2008 i686 GNU/Linux
> >> >
> >> >
> >> > I used a script to check the interupts
> >> >
> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> > LOC:   14892854   14892843   14892841   14892830
> >> > LOC:   14892955   14892944   14892942   14892931
> >> > LOC:   14893056   14893045   14893043   14893032
> >> > LOC:   14893157   14893146   14893144   14893133
> >> > LOC:   14893258   14893247   14893245   14893234
> >> > LOC:   14893359   14893348   14893346   14893335
> >> > LOC:   14893460   14893449   14893447   14893436
> >> >
> >> >
> >> > 2) run dingaling with client mode. dingaling doesn't pick up answer. I
> >> > originate a call from FS, the console blocks, gtalk client ring and it
> >> > looks ok when I hit answer, but no sound, FS console still blocks. a
> >> > few seconds later gtalk client shows:
> >> >
> >> > Sorry! The voice chat with language_lesson failed because of a network
> >> > problem at 5:50 PM. Please try again.
> >> >  Click here to report this to Google.
> >> >
> >> >
> >> > here is a log with bgapi:
> >> > http://pastebin.freeswitch.org/12434
> >> >
> >> >
> >> > rev 14696 works on the same server same conf. another server running
> >> > 16958M works at the same time. I have no clue to check, is it related
> >> > to the timer as in 1)? can some help me to take a look? let me know if
> >> > you need "dl_debug on" logs.
> >> >
> >> > Thanks.
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar 15 10:21:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 11:21:48 -0600
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268672644810-4738333.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
Message-ID: <191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>

As you can see in the log that call hangs up before you bridge it so it ends
in failure
you may want to wait and make sure the call makes it to park start instead
of answered before you bridge.


On Mon, Mar 15, 2010 at 11:04 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hi Anthony,
> Sorry if this was unclear - I am actually waiting for the phone to be
> answered. The two calls are bridged in handler for ANSWERED event of the
> 2nd
> call...
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4738333.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Mar 15 10:36:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 09:36:03 -0800
Subject: [Freeswitch-users] problem in sending calls to internal Gateway.
In-Reply-To: <249845.71206.qm@web65402.mail.ac4.yahoo.com>
References: <249845.71206.qm@web65402.mail.ac4.yahoo.com>
Message-ID: <87f2f3b91003151036s3d458580l91c4cec4d90d3e73@mail.gmail.com>

Prabin,

Thank you for posting the configs. It is difficult to say what is wrong at
this point because we don't know exactly what you are dialing. Could you
please use pastebin.freeswitch.org and give us some call examples? Capture
the debug output from the FreeSWITCH console. Also, be sure to say what
phone is making the call and what digits the phone is dialing. You might
actually be able to figure it out by looking at the debug output. If you are
using fs_cli then debug mode is on automatically. If you are at the actual
FreeSWITCH console (meaning that you did NOT start freeswitch with the -nc
flag) then be sure to press F8 or type "console loglevel debug" to enable
debug level output. See also this page for lots of handy tips on reporting
information for troubleshooting:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Sat, Mar 13, 2010 at 10:22 PM, Prabin Shrestha <nepaligas at yahoo.com>wrote:

> hello,
>
> My extension are working properly. But they can't send calls to gateway. I
> have 2 interfaces, one with public and another with private lan. Gateway is
> in private lan. Please suggest.
> My configurations are as follows:
>
> /conf/sip_profiles/external.xml:
>    <param name="rtp-ip" value="110.4.12.111"/>
>    <param name="sip-ip" value="110.4.12.111"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>
>
> /conf/sip_profiles/internal.xml:
>     <param name="rtp-ip" value="192.168.1.11"/>
>     <param name="sip-ip" value="192.168.1.11"/>
>    <param name="hold-music" value="$${hold_music}"/>
>    <param name="apply-nat-acl" value="nat.auto"/>
>
> haven't made any changes in conf/autoload_configs/acl.conf.xml
>
> to route calls from providers, /dialplan/public/providers.xml:
>    <extension name="provider">
>    <condition field="network_addr" expression="200.144.44.177"/>
>    <condition field="destination_number" expression="^(\d+)$">
>        <action application="transfer" data="$1 XML context_AX"/>
>    </condition>
>   </extension>
>
> gateway is defined at /conf/sip_profiles/internal/gateways.xml:
>   <include>
>   <gateway name="gateway_AX">
>    <param name="username" value="admin"/>
>    <param name="password" value="nepal123"/>
>    <param name="proxy" value="192.168.1.100"/>
>    <param name="register" value="false"/>
>    <param name="extension" value="1001"/>
>    </gateway>
>   </include>
>
>
> To route internal calls from extensions, /conf/dialplan/default.xml:
> <extension name="my_internal">
>    <!--<condition field="${ruleset}" expression="internal" /> -->
>    <condition field="destination_number" expression="^(d+)$">
>      <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
>    </condition>
>  </extension>
>
> to route calls from providers, conf/dialplan/inbound_routing.xml:
> <context name="context_AX">
>    <extension name="from_provider_to_AX">
>    <!--<condition field="destination_number" expression="^00(1.*)$">-->
>    <condition field="destination_number" expression="^9(\d+)$">
>    <action application="bridge" data="sofia/gateway/gateway_AX/$1"/>
>    </condition>
>    </extension>
> </context>
>
> Please help.
>
>
>      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage.
> http://in.yahoo.com/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From robert.dyck at shaw.ca  Sun Mar 14 18:11:57 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Sun, 14 Mar 2010 18:11:57 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
Message-ID: <201003141811.58645.robert.dyck@shaw.ca>

16987 on Fedora

I realize fscomm is an immature project. I get an error message when it tries 
to detect my devices ( pci sound card, usb handset, usb camera with microphone 
). Is this the expected result? Detection works in Twinkle and Ekiga.

Second question - is video support on the roadmap?



From schogge at gmx.de  Mon Mar 15 08:14:02 2010
From: schogge at gmx.de (schogge)
Date: Mon, 15 Mar 2010 08:14:02 -0700 (PDT)
Subject: [Freeswitch-users]  FreeSWITCH to add SRTP to Asterisk
Message-ID: <27901964.post@talk.nabble.com>


Hi Community,
I'm new to FreeSWITCH. What I have read so far it seems to be a very
interesting project.
For our company we are running an Asterisk PBX in a datacenter. After some
years it is doing everything we want except of SRTP.
Would it be possible to run FreeSWITCH in front of Asterisk to convert SRTP
coming from our phones to RTP which is ok for Asterisk and vice versa?
Thanks,
schogge
-- 
View this message in context: http://old.nabble.com/FreeSWITCH-to-add-SRTP-to-Asterisk-tp27901964p27901964.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From infos at madovsky.org  Mon Mar 15 09:28:29 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 12:28:29 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>

Hi all,

is there any way to set the voicemail for leave a message without any option at the end ?
like record or send the message by email once the caller hangs up or after N sec of silence.

Thanks

Franck
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From anthony.minessale at gmail.com  Mon Mar 15 10:54:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 12:54:57 -0500
Subject: [Freeswitch-users] FreeSWITCH to add SRTP to Asterisk
In-Reply-To: <27901964.post@talk.nabble.com>
References: <27901964.post@talk.nabble.com>
Message-ID: <191c3a031003151054s28d84c38oc02ae3ddad38acf7@mail.gmail.com>

yes, it should be possible.


On Mon, Mar 15, 2010 at 10:14 AM, schogge <schogge at gmx.de> wrote:

>
> Hi Community,
> I'm new to FreeSWITCH. What I have read so far it seems to be a very
> interesting project.
> For our company we are running an Asterisk PBX in a datacenter. After some
> years it is doing everything we want except of SRTP.
> Would it be possible to run FreeSWITCH in front of Asterisk to convert SRTP
> coming from our phones to RTP which is ok for Asterisk and vice versa?
> Thanks,
> schogge
> --
> View this message in context:
> http://old.nabble.com/FreeSWITCH-to-add-SRTP-to-Asterisk-tp27901964p27901964.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From spencer at 5ninesolutions.com  Mon Mar 15 11:07:24 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 15 Mar 2010 11:07:24 -0700
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
	<191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
Message-ID: <5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>

I'm using Centos 5.4, I have a build machine and I got the latest  
source from SVN, and rebuilt the RPMS from the updates source and then  
updated the FS machine.  We have our own repo setup to update the  
machines.  I disabled most of the modules except mod_console,  
mod_logfile, mod_cdr_csv, mod_event_scoket, mod_sofia, mod_loopback,  
mod_commands, mod_dptools, mod_expr, mod_limit, mod_dialplan_xml, and  
mod_voipcodecs and everything is working great.  As soon as I enable  
the default set of modules, FS will work for about 3-4 hours and then  
one of the threads goes to 100% CPU.  I haven't had the time to load  
the modules one by one to try to isolate the problem.  The build  
completes fine, and I'm using the stock centos libraries to build  
against.  The centos xen kernel on x86_64 is 100hz, could that be an  
issue?


On Mar 15, 2010, at 8:45 AM, Anthony Minessale wrote:

> That seems suspiciously similar to an old issue we have already  
> fixed, how exactly did you rebuild?
>
>
> On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> I rebuilt today.
>
> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>
> > Are you sure you're on SVN trunk?
> >
> > /b
> >
> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
> >
> >> No.  Its on our hardware, we just use xen to virtualize the
> >> machines.. this is the only vm on the machine.  Gcore complains
> >> about no debugging symbols found, do I need to rebuild?
> >>
> >> It appears to be a problem when registering to one of my trunks.
> >> The very strange thing is the everything will be working perfectly
> >> and then all of a sudden this happens, i.e. trunks regged, calls
> >> completing, etc. This is a very lightly loaded server.
> >>
> >> This is the output from the console:
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Mon Mar 15 11:18:39 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:18:39 -0800
Subject: [Freeswitch-users] Dial number at registered extension
In-Reply-To: <ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>
References: <CC765174-97CF-47E1-80AE-CBFC05EE2903@5ninesolutions.com>
	<b39273531003141412j799a9c00td7f08d143a835070@mail.gmail.com>
	<AD7C61B4-A2CC-4E49-966F-542A070F8B87@5ninesolutions.com>
	<2FA6B68D-360A-4E03-A1CD-C7A3CF7FC97C@freeswitch.org>
	<03BD849C-2B7F-4A59-AAB9-15E977EDD207@5ninesolutions.com>
	<ebbc2d551003141617o3e949c18j9a983d9e032ac816@mail.gmail.com>
	<10061342-9B60-427D-979F-B3F762D1AEAC@5ninesolutions.com>
	<3A691E53-08FE-4897-9896-225A823934D0@5ninesolutions.com>
	<ebbc2d551003141940l549b25f5x2110862560519498@mail.gmail.com>
Message-ID: <87f2f3b91003151118u1a0373aete4903979e48aaab8@mail.gmail.com>

FYI,
I added this as example 16 on the dialplan xml page on the wiki.
-MC


2010/3/14 Jo?o Mesquita <jmesquita at freeswitch.org>

> Wiki is your friend to that type of info! Contribute to it, please!
>
> JM
>
>
>
> On Sun, Mar 14, 2010 at 11:20 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> So just in case anyone else ever needs to do this.. here's what I ended up
>> with:
>>
>> <include>
>>   <extension name="NNNNNNNNNNN-did">
>>     <condition field="destination_number" expression="^(NNNNNNNNNNN)$">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="set" data="accountcode=<ACCTNUM>"/>
>>       <action application="bridge"
>> data="sofia/default/$1${regex(${sofia_contact(${accountcode}@$${domain})}|^[^\@]+(.*)|%1)}"
>> />
>>     </condition>
>>   </extension>
>> </include>
>>
>> Basically the extension UAs register to is created as the same as the
>> account code.  We then use their ip at registration to dial a did on their
>> end.
>>
>> Spencer
>>
>>
>> On Mar 14, 2010, at 5:25 PM, Spencer Thomason wrote:
>>
>> Duh.. :-)  Thanks!
>>
>> On Mar 14, 2010, at 4:17 PM, Jo?o Mesquita wrote:
>>
>> Yes, sofia_contact with regex to extract the IP portion? You had the
>> answer yourself.
>>
>> Jo?o Mesquita
>>
>>
>>
>> On Sun, Mar 14, 2010 at 8:00 PM, Spencer Thomason <
>> spencer at 5ninesolutions.com> wrote:
>>
>>> I guess my question is, is there some function which will return the
>>> IP address of a registered user?  Similar to sofia_contact but only
>>> the IP address?
>>>
>>> Spencer
>>>
>>> On Mar 14, 2010, at 3:18 PM, Brian West wrote:
>>>
>>> > You sure can still have it call a registration.. You're mixing the
>>> > dialplan and user directly as if they are a single entity.  They
>>> > aren't.
>>> >
>>> > /b
>>> >
>>> > On Mar 14, 2010, at 5:02 PM, Spencer Thomason wrote:
>>> >
>>> >> That looks promising, thanks.  I was hoping to have FS do it
>>> >> automatically.  I.e. inbound dids are assigned to specific asterisk
>>> >> boxes, ITSP dials inbound did of freeswitch, freeswitch dials
>>> >> inbound did @ whereever the registration came from instead of
>>> >> having the IP address hard coded into a route.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Mar 15 11:22:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Mar 2010 13:22:59 -0500
Subject: [Freeswitch-users] 100% CPU
In-Reply-To: <5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>
References: <E7527BE7-8263-4BC3-8A5C-0D4C21B624DB@5ninesolutions.com>
	<A18CC174-AD98-4867-9623-F7494324A559@freeswitch.org>
	<354A4EF2-6A17-4153-B500-65EDB00A4D4D@5ninesolutions.com>
	<191c3a031003141608j4933d768p116b17c32be2b6d2@mail.gmail.com>
	<E2C36AEB-1D2F-4F22-BD5B-EFB90691B118@5ninesolutions.com>
	<F677255B-1E66-4D63-84ED-A00462D16CEE@freeswitch.org>
	<C42CAC9D-E7CB-4E62-BBCF-0D986BB230A1@5ninesolutions.com>
	<191c3a031003150845y63b03bf8ic46b5ba5b8d830bd@mail.gmail.com>
	<5D46FB05-7C47-4D87-8CBB-1B1FEEC64A87@5ninesolutions.com>
Message-ID: <191c3a031003151122q601b8341s1f77172090ebc02f@mail.gmail.com>

if you can email me privately at consulting at freeswitch.org with the details
to log into the box.
I can have a look for you.


On Mon, Mar 15, 2010 at 1:07 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> I'm using Centos 5.4, I have a build machine and I got the latest source
> from SVN, and rebuilt the RPMS from the updates source and then updated the
> FS machine.  We have our own repo setup to update the machines.  I disabled
> most of the modules except mod_console, mod_logfile, mod_cdr_csv,
> mod_event_scoket, mod_sofia, mod_loopback, mod_commands, mod_dptools,
> mod_expr, mod_limit, mod_dialplan_xml, and mod_voipcodecs and everything is
> working great.  As soon as I enable the default set of modules, FS will work
> for about 3-4 hours and then one of the threads goes to 100% CPU.  I haven't
> had the time to load the modules one by one to try to isolate the problem.
>  The build completes fine, and I'm using the stock centos libraries to build
> against.  The centos xen kernel on x86_64 is 100hz, could that be an issue?
>
>
> On Mar 15, 2010, at 8:45 AM, Anthony Minessale wrote:
>
> That seems suspiciously similar to an old issue we have already fixed, how
> exactly did you rebuild?
>
>
> On Sun, Mar 14, 2010 at 6:38 PM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> I rebuilt today.
>>
>> On Mar 14, 2010, at 5:23 PM, Brian West wrote:
>>
>> > Are you sure you're on SVN trunk?
>> >
>> > /b
>> >
>> > On Mar 14, 2010, at 7:14 PM, Spencer Thomason wrote:
>> >
>> >> No.  Its on our hardware, we just use xen to virtualize the
>> >> machines.. this is the only vm on the machine.  Gcore complains
>> >> about no debugging symbols found, do I need to rebuild?
>> >>
>> >> It appears to be a problem when registering to one of my trunks.
>> >> The very strange thing is the everything will be working perfectly
>> >> and then all of a sudden this happens, i.e. trunks regged, calls
>> >> completing, etc. This is a very lightly loaded server.
>> >>
>> >> This is the output from the console:
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Mar 15 11:23:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:23:02 -0800
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
Message-ID: <87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>

On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi all,
>
> is there any way to set the voicemail for leave a message without any
> option at the end ?
> like record or send the message by email once the caller hangs up or after
> N sec of silence.
>
>
I'm not quite sure that I understand the question. If the caller hangs up
while leaving a message then the message gets sent.
-MC
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From msc at freeswitch.org  Mon Mar 15 11:28:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:28:35 -0800
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <8259813661645760166@unknownmsgid>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
	<8259813661645760166@unknownmsgid>
Message-ID: <87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>wrote:

>  Hi
>
>
>
> I have now configured the gateways as suggested below but I am not able to
> see the sip_gateway_name variable when the call hangup event is complete. I
> am on Freeswitch 1.0.5.
>
>
>
Where are you looking to find the variable?
-MC
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From msc at freeswitch.org  Mon Mar 15 11:39:01 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:39:01 -0800
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <87f2f3b91003151139w5525dfa9pcb1d201bdb0ee91d@mail.gmail.com>

On Mon, Mar 15, 2010 at 7:36 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
> like just farwading the call.
>

1.0.2? Are you running it on MS-DOS 2.0?!
Seriously, you need to update. 1.0.2 had some bugs. We're now at 1.0.5
(essentially). 1.0.2 is like 3 generations old.

> <action application="bridge" data=<data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>
> sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
> />
> i do not using any gateway here, just farwarding call to router, is this
> correct way of doing? any idea
>

You might want to use $1 instead of ${destination_number} since $1 will
always have the right information in it, assuming of course that you used (
and ) in your regular expression. Something like this ought to work

<action application="bridge" data="sofia/internal/$1 at 1.2.3.4"/> <!-- 1.2.3.4
is the gateway IP addr -->

-MC
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From lloyd.aloysius at gmail.com  Mon Mar 15 11:50:25 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 14:50:25 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <201003151102.03407.sos@sokhapkin.dyndns.org>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
Message-ID: <8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>

How to control the default-template. I do not want to run all the time when
call hangup. I want only when I need for some hangup information..

Thanks
Lloyd


On Mon, Mar 15, 2010 at 11:02 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> That script will be executed automatically when call ends. You can set some
> channel variables in the main lua script and access those variables from
> hangup lua script if you need to pass the information to the hangup script.
>
> On Monday 15 March 2010, Aloysius Lloyd wrote:
> > How to trigger this template from the main script ? or Does this
> > automatically trigger when finish the first script complete?
> >
> > Thanks
> > Lloyd
> >
> > On Mon, Mar 15, 2010 at 10:48 AM, Sergey Okhapkin
> >
> > <sos at sokhapkin.dyndns.org>wrote:
> > > cdr_csv.conf.xml:
> > >
> > > <param name="default-template" value="lua"/>
> > > <templates>
> > >  <template name="lua">${lua(onhangup.lua)}</template>
> > >
> > > All variables described at http://wiki.freeswitch.org/wiki/Mod_cdr_csv
> > > are accessible from that lua script.
> > >
> > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > How to execute another lua script when channel enter the REPORTING
> > > > state?
> > > >
> > > > What are the channel variable names?
> > > >
> > > > Thanks
> > > > Lloyd
> > > >
> > > > On Mon, Mar 15, 2010 at 9:56 AM, Sergey Okhapkin
> > > >
> > > > <sos at sokhapkin.dyndns.org>wrote:
> > > > > Unfortunately all "interesting" billing-related channel variables
> are
> > >
> > > set
> > >
> > > > > after the lua script execution, when channel enters REPORTING
> state.
> > > > > To access
> > > > > those variables I execute separate lua script using mod_cdr_csv, it
> > > > > allows to
> > > > > invoke lua script.
> > > > >
> > > > > On Monday 15 March 2010, Aloysius Lloyd wrote:
> > > > > > Hi All,
> > > > > >
> > > > > > I have the following simple Lua script answer a inbound call then
> > >
> > > dial
> > >
> > > > > > another number and bridge the call.
> > > > > >
> > > > > > Is there any way to get the  call duration or billing sec using
> the
> > > > > >
> > > > > > session:getVariable(  );
> > > > > >
> > > > > >
> > > > > >
> > > > > > -- answer the call
> > > > > > session:answer();
> > > > > >
> > > > > > -- sleep a second
> > > > > > session:sleep(1000);
> > > > > >
> > > > > > -- Initiate an outbound call
> > > > > > obSession =
> > > > >
> > > > > freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> > > > >
> > > > > > obSession:execute("sched_hangup","+60 alloted_timeout");
> > > > > >
> > > > > > -- bride the call
> > > > > >     freeswitch.bridge(session, obSession);
> > > > > >
> > > > > > -- hangup
> > > > > > session:hangup();
> > > > > >
> > > > > >
> > > > > > Thank you
> > > > > > Lloyd
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > >
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> >
>
>
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From msc at freeswitch.org  Mon Mar 15 11:52:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 10:52:31 -0800
Subject: [Freeswitch-users] FreeSWITCH Conf Call Agenda
Message-ID: <87f2f3b91003151152s1acc22f4ydf45f2e6630673b3@mail.gmail.com>

Hello all,

The agenda for Wednesday's conference call is now up:
http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_17

We could use some more content. Also, we would love to have a volunteer
discuss one of the FreeSWITCH modules or some other FS feature that he/she
uses. If you have any thoughts or suggestions please email me off list.

Thanks!
-MC
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From infos at madovsky.org  Mon Mar 15 12:18:10 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 15:18:10 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
Message-ID: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>

Actually on the default voicemail example 
after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
so if the caller hangs up before to press 2 the message is not sent. any way to remove
these options and send the message after slience or caller hang up ?

Thanks

F
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 2:23 PM
  Subject: Re: [Freeswitch-users] voicemail options





  On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

    Hi all,

    is there any way to set the voicemail for leave a message without any option at the end ?
    like record or send the message by email once the caller hangs up or after N sec of silence.

  I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
  -MC





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From testeador01 at gmail.com  Mon Mar 15 12:32:04 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 15 Mar 2010 14:32:04 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
Message-ID: <c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>

Hello,

I have voicemails set on my office and messages do get sent when you hang up
even if you don't press 2,

did you try it? or are you guessing? (not trying to be rude, just trying to
get info :) )

what revision are you running?



2010/3/15 Madovsky <infos at madovsky.org>

>  Actually on the default voicemail example
> after you left a message the operator says press 1 to listen blabla or 2 to
> save blabla..
> so if the caller hangs up before to press 2 the message is not sent. any
> way to remove
> these options and send the message after slience or caller hang up ?
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 15, 2010 2:23 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
>
>
> On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  Hi all,
>>
>> is there any way to set the voicemail for leave a message without any
>> option at the end ?
>> like record or send the message by email once the caller hangs up or after
>> N sec of silence.
>>
>>
> I'm not quite sure that I understand the question. If the caller hangs up
> while leaving a message then the message gets sent.
> -MC
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon Mar 15 13:01:38 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:01:38 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
Message-ID: <C3263B8079914E40A8653B84DA785598@MOBILEE1705>

Hi,

ah ok, so maybe the reason of fail is this error in console

2010-03-15 16:01:36.752887 [CRIT] switch_channel.c:759 Invalid data (${RECORD_COMMENT} contains a variable)

Any idea ?

Thanks
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 3:32 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


  did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


  what revision are you running?






  2010/3/15 Madovsky <infos at madovsky.org>

    Actually on the default voicemail example 
    after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
    so if the caller hangs up before to press 2 the message is not sent. any way to remove
    these options and send the message after slience or caller hang up ?

    Thanks

    F
      ----- Original Message ----- 
      From: Michael Collins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 2:23 PM
      Subject: Re: [Freeswitch-users] voicemail options





      On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

        Hi all,

        is there any way to set the voicemail for leave a message without any option at the end ?
        like record or send the message by email once the caller hangs up or after N sec of silence.

      I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
      -MC





--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 13:02:05 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:02:05 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
Message-ID: <A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>

Sorry revision 16970
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 3:32 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


  did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


  what revision are you running?






  2010/3/15 Madovsky <infos at madovsky.org>

    Actually on the default voicemail example 
    after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
    so if the caller hangs up before to press 2 the message is not sent. any way to remove
    these options and send the message after slience or caller hang up ?

    Thanks

    F
      ----- Original Message ----- 
      From: Michael Collins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 2:23 PM
      Subject: Re: [Freeswitch-users] voicemail options





      On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

        Hi all,

        is there any way to set the voicemail for leave a message without any option at the end ?
        like record or send the message by email once the caller hangs up or after N sec of silence.

      I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
      -MC





--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From testeador01 at gmail.com  Mon Mar 15 13:10:54 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 15 Mar 2010 15:10:54 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705>
	<87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com>
	<A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
Message-ID: <c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>

Hello,

What value are you setting for "RECORD_COMMENT"? what is your dial plan for
the extension you're testing?



2010/3/15 Madovsky <infos at madovsky.org>

>  Sorry revision 16970
>
> ----- Original Message -----
> *From:* Milena <testeador01 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 15, 2010 3:32 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
> Hello,
>
> I have voicemails set on my office and messages do get sent when you hang
> up even if you don't press 2,
>
> did you try it? or are you guessing? (not trying to be rude, just trying to
> get info :) )
>
> what revision are you running?
>
>
>
> 2010/3/15 Madovsky <infos at madovsky.org>
>
>>  Actually on the default voicemail example
>> after you left a message the operator says press 1 to listen blabla or 2
>> to save blabla..
>> so if the caller hangs up before to press 2 the message is not sent. any
>> way to remove
>> these options and send the message after slience or caller hang up ?
>>
>> Thanks
>>
>> F
>>
>>   ----- Original Message -----
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Monday, March 15, 2010 2:23 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>>
>>
>> On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  Hi all,
>>>
>>> is there any way to set the voicemail for leave a message without any
>>> option at the end ?
>>> like record or send the message by email once the caller hangs up or
>>> after N sec of silence.
>>>
>>>
>> I'm not quite sure that I understand the question. If the caller hangs up
>> while leaving a message then the message gets sent.
>> -MC
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From xengelpublicx at gmail.com  Mon Mar 15 13:21:07 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Mon, 15 Mar 2010 23:21:07 +0300
Subject: [Freeswitch-users] intercept in fifo queue
Message-ID: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>

Hello.

How can I intercept a fifo? I tried to use the example of intecept
default dialplan, but in no fifo dialed_extension.

    <extension name="operators">
      <condition field="destination_number" expression="^operators$">
            <action application="bind_meta_app" data="2 b s
record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
            <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
            <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
            <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
            <action application="answer"/>
            <action application="playback"
data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
            <action application="playback"
data="ivr/ivr-hold_connect_call.wav"/>
            <action application="set" data="fifo_music=$${hold_music}"/>
            <action application="fifo" data="operators@$${domain} in undef"/>
      </condition>
    </extension>

    <fifo name="operators@$${domain}" importance="0">
      <member timeout="15" simo="1"
lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member>
    </fifo>
  </fifos>
-- 
Best regards, Vladimir Elizarov



From robert.hadley at teotech.com  Mon Mar 15 13:24:07 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 15 Mar 2010 13:24:07 -0700
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com>
	<9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
Message-ID: <6CFB240D4D064C63BB295CAADAD1EF9E@greyhawk.tonecommander.com>

 

The default FS action is to save the voicemail, so it is saved when there is
silence or the caller hangs up without pressing 2.  Note there is a minimum
recording length of several seconds before a VM is saved.

 

Robert

 

  _____  

From: Madovsky [mailto:infos at madovsky.org] 
Sent: Monday, March 15, 2010 12:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] voicemail options

 

Actually on the default voicemail example 

after you left a message the operator says press 1 to listen blabla or 2 to
save blabla..

so if the caller hangs up before to press 2 the message is not sent. any way
to remove

these options and send the message after slience or caller hang up ?

 

Thanks

 

F

----- Original Message ----- 

From: Michael Collins <mailto:msc at freeswitch.org>  

To: freeswitch-users at lists.freeswitch.org 

Sent: Monday, March 15, 2010 2:23 PM

Subject: Re: [Freeswitch-users] voicemail options

 

 

On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

Hi all,

 

is there any way to set the voicemail for leave a message without any option
at the end ?

like record or send the message by email once the caller hangs up or after N
sec of silence.

 

I'm not quite sure that I understand the question. If the caller hangs up
while leaving a message then the message gets sent. 
-MC

 


  _____  


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 13:24:56 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:24:56 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
Message-ID: <5B1C9610DC644E8390EA3B9D450F4DBB@MOBILEE1705>

 here is my dialplan

<extension name="phone_connection" continue="true">
   <condition field="destination_number" expression="^(123456789)$">
   <action application="set" data="dialed_extension=$1"/>
   <action application="export" data="dialed_extension=$1"/>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail=true"/>
   <action application="set" data="ignore_early_media=true"/>
   <action application="set" data="call_timeout=15"/>
   <action application="bridge" data="{monitor_early_media_fail=user_busy:2:480+620!user_not_registered:2:1776.7!subscriber_absent:2:1776.7,!no_user_response:2:1776.7,sip_from_uri=${sip_from_user}@${domain}}user/${dialed_extension}@${domain}"/>
   <action application="set" data="call_timeout=15"/>
   <action application="set" data="originate_timeout=5"/>
   <action application="bridge" data="{monitor_early_media_fail=user_busy:2:480+620!user_not_registered:2:1776.7!subscriber_absent:2:1776.7,!no_user_response:2:1776.7,origination_caller_id_name=${outbound_caller_id_name},origination_caller_id_number=${outbound_caller_id}}sofia/gateway/${distributor(pstn_domestic)}/00${destination_number}|sofia/gateway/${distributor(pstn_international_2)}/011${destination_number}" loop="2"/>
   <action application="answer"/>
   <action application="sleep" data="1000"/>
  <action application="voicemail" data="default ${domain} ${dialed_extension}"/>

absolutly have no idea where RECORD_COMMENT var comes from...

Thx
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  What value are you setting for "RECORD_COMMENT"? what is your dial plan for the extension you're testing?






  2010/3/15 Madovsky <infos at madovsky.org>

    Sorry revision 16970
      ----- Original Message ----- 
      From: Milena 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 3:32 PM
      Subject: Re: [Freeswitch-users] voicemail options


      Hello,


      I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


      did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


      what revision are you running?






      2010/3/15 Madovsky <infos at madovsky.org>

        Actually on the default voicemail example 
        after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
        so if the caller hangs up before to press 2 the message is not sent. any way to remove
        these options and send the message after slience or caller hang up ?

        Thanks

        F
          ----- Original Message ----- 
          From: Michael Collins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 2:23 PM
          Subject: Re: [Freeswitch-users] voicemail options





          On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

            Hi all,

            is there any way to set the voicemail for leave a message without any option at the end ?
            like record or send the message by email once the caller hangs up or after N sec of silence.

          I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
          -MC





----------------------------------------------------------------------


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From infos at madovsky.org  Mon Mar 15 13:38:42 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:38:42 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705>
	<6CFB240D4D064C63BB295CAADAD1EF9E@greyhawk.tonecommander.com>
Message-ID: <AE28A09C45E94D99804FBD5B5256E738@MOBILEE1705>

yes by

      <param name="min-record-len" value="5"/>
      <param name="max-record-len" value="180"/>

was not sure that this save behavior was in default FS action.

Thanks

F
  ----- Original Message ----- 
  From: Robert Hadley 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:24 PM
  Subject: Re: [Freeswitch-users] voicemail options


   

  The default FS action is to save the voicemail, so it is saved when there is silence or the caller hangs up without pressing 2.  Note there is a minimum recording length of several seconds before a VM is saved.

   

  Robert

   


------------------------------------------------------------------------------

  From: Madovsky [mailto:infos at madovsky.org] 
  Sent: Monday, March 15, 2010 12:18 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] voicemail options

   

  Actually on the default voicemail example 

  after you left a message the operator says press 1 to listen blabla or 2 to save blabla..

  so if the caller hangs up before to press 2 the message is not sent. any way to remove

  these options and send the message after slience or caller hang up ?

   

  Thanks

   

  F

    ----- Original Message ----- 

    From: Michael Collins 

    To: freeswitch-users at lists.freeswitch.org 

    Sent: Monday, March 15, 2010 2:23 PM

    Subject: Re: [Freeswitch-users] voicemail options

     

     

    On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

    Hi all,

     

    is there any way to set the voicemail for leave a message without any option at the end ?

    like record or send the message by email once the caller hangs up or after N sec of silence.

     

    I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
    -MC

     


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From infos at madovsky.org  Mon Mar 15 13:59:37 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 16:59:37 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
Message-ID: <180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>

ok maybe this is the var from voicemail.conf.xml record-comment ?
      <param name="record-comment" value="message from a call at ${domain}"/>
didn't know it doesn't allow variables... so I removed it and the error disappeared.
so I have on console

2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz

but no message sent by email yet..
I checked sendmail -t manually, it works.

here is my voicemail.conf.xml

<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="mp3"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="5"/>
      <param name="max-record-len" value="180"/>
      <param name="max-retries" value="3"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="login-keys" value="0"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="skip-greet-key" value="#"/>
      <!--<param name="previous-message-key" value=""/>-->
      <!--<param name="next-message-key" value=""/>-->
      <!--<param name="skip-info-key" value=""/>-->
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <param name="storage-dir" value="/tmp"/>
      <param name="odbc-dsn" value="freeswitch-pgsql:freeswitch:bronto8fc"/>
      <param name="record-comment" value="message from a call"/>
      <param name="record-title" value="your Boophone voicemail"/>
      <param name="record-copyright" value="own Copyright"/>
    </profile>
  </profiles>
</configuration>


and the user config
....
    <params>
      ....
     <param name="vm-email-all-messages" value="true"/>
     <param name="vm-attach-file" value="true"/>
     <param name="vm-keep-local-after-email" value="false"/>
      <param name="vm-mailto" value="infos at baba.bibi"/>
...
    </params>


F



  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Hello,


  What value are you setting for "RECORD_COMMENT"? what is your dial plan for the extension you're testing?






  2010/3/15 Madovsky <infos at madovsky.org>

    Sorry revision 16970
      ----- Original Message ----- 
      From: Milena 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 3:32 PM
      Subject: Re: [Freeswitch-users] voicemail options


      Hello,


      I have voicemails set on my office and messages do get sent when you hang up even if you don't press 2, 


      did you try it? or are you guessing? (not trying to be rude, just trying to get info :) ) 


      what revision are you running?






      2010/3/15 Madovsky <infos at madovsky.org>

        Actually on the default voicemail example 
        after you left a message the operator says press 1 to listen blabla or 2 to save blabla..
        so if the caller hangs up before to press 2 the message is not sent. any way to remove
        these options and send the message after slience or caller hang up ?

        Thanks

        F
          ----- Original Message ----- 
          From: Michael Collins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 2:23 PM
          Subject: Re: [Freeswitch-users] voicemail options





          On Mon, Mar 15, 2010 at 8:28 AM, Madovsky <infos at madovsky.org> wrote:

            Hi all,

            is there any way to set the voicemail for leave a message without any option at the end ?
            like record or send the message by email once the caller hangs up or after N sec of silence.

          I'm not quite sure that I understand the question. If the caller hangs up while leaving a message then the message gets sent. 
          -MC





----------------------------------------------------------------------


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          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 14:14:52 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 16:14:52 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705>
	<c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com>
	<180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
Message-ID: <2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>

Word of advice.  MP3 is not recommended at all.

/b

On Mar 15, 2010, at 3:59 PM, Madovsky wrote:

> 2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
> 2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz

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From msc at freeswitch.org  Mon Mar 15 14:15:03 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Mar 2010 13:15:03 -0800
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
Message-ID: <87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>

On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

> How to control the default-template. I do not want to run all the time when
> call hangup. I want only when I need for some hangup information..
>
> Thanks
> Lloyd
>

I don't know that you can selectively run or not run a script at hangup
time. However, since Lua is so lightweight it probably would be fine just to
run it at each hangup.
-MC
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From lloyd.aloysius at gmail.com  Mon Mar 15 14:38:24 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 15 Mar 2010 17:38:24 -0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
	<201003151048.34689.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>
	<201003151102.03407.sos@sokhapkin.dyndns.org>
	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>
	<87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
Message-ID: <8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>

I would like to do some Database insert and update operations , Not for all
the calls only the calls originating from a particular script.







On Mon, Mar 15, 2010 at 5:15 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>> How to control the default-template. I do not want to run all the time
>> when call hangup. I want only when I need for some hangup information..
>>
>> Thanks
>> Lloyd
>>
>
> I don't know that you can selectively run or not run a script at hangup
> time. However, since Lua is so lightweight it probably would be fine just to
> run it at each hangup.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From nazim.agabekov at gmail.com  Mon Mar 15 15:09:12 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Tue, 16 Mar 2010 02:09:12 +0400
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>	<201003151048.34689.sos@sokhapkin.dyndns.org>	<8a19bf2e1003150754m558f72d3jbdad26217d24e82a@mail.gmail.com>	<201003151102.03407.sos@sokhapkin.dyndns.org>	<8a19bf2e1003151150t70220eefy6e2281abfdbd4a81@mail.gmail.com>	<87f2f3b91003151415t48a342c0wb6c5e81f42bf734c@mail.gmail.com>
	<8a19bf2e1003151438t532b522lb9f98462d744317@mail.gmail.com>
Message-ID: <4B9EB008.4080005@gmail.com>

Lloyd, if you need to run different Lua scripts depending on some
variables (or entirely skip the cdr), try my implementation.
Documentation is in very embryonic state. You could get the overview
from blog.buta-tech.com. Warning! all code snapshots on that page are
obsolete! Latetest svn trunk is at
http://nazim at svn.freeswitch.org/svn/freeswitch/trunk/contrib/nazim

Regards,
Nazim


On 03/16/2010 01:38 AM, Aloysius Lloyd wrote:
> I would like to do some Database insert and update operations , Not
> for all the calls only the calls originating from a particular script.
>
>
>
>
>
>
>
> On Mon, Mar 15, 2010 at 5:15 PM, Michael Collins <msc at freeswitch.org
> <mailto:msc at freeswitch.org>> wrote:
>
>
>
>     On Mon, Mar 15, 2010 at 10:50 AM, Aloysius Lloyd
>     <lloyd.aloysius at gmail.com <mailto:lloyd.aloysius at gmail.com>> wrote:
>
>         How to control the default-template. I do not want to run all
>         the time when call hangup. I want only when I need for some
>         hangup information..
>
>         Thanks
>         Lloyd
>
>
>     I don't know that you can selectively run or not run a script at
>     hangup time. However, since Lua is so lightweight it probably
>     would be fine just to run it at each hangup.
>     -MC
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From will.traenkle at yahoo.com  Mon Mar 15 16:22:25 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Mon, 15 Mar 2010 16:22:25 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination fails
	// direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
Message-ID: <923176.22123.qm@web57601.mail.re1.yahoo.com>

When a hunt group is accessed from the main IVR, the hunt group dials
an internal extension via correctly but when trying to dial an
external number such a cell phone via sip uri it is unable to connect
or ring the cell phone and it waits the correct timeout period before it
goes to voicemail.

However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  


There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.

1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR

2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3

3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3

4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
 
5) Dialplan default.xml:  
http://pastebin.com/iAbNSW9E 
 
If you require additional information or have any other
questions, please let me know.

Thanks,

-Will

________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, March 9, 2010 11:00:53 PM
Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds




On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>
>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>
>Any ideas where I should look and what line of code I might be missing?
>
>

Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
-MC


      
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From lloydie.t at googlemail.com  Mon Mar 15 16:51:48 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Mon, 15 Mar 2010 23:51:48 +0000
Subject: [Freeswitch-users] Problem post request.
Message-ID: <8b61bd671003151651sda3970er7e2d2fcf50067ae2@mail.gmail.com>

I post two request re mod_xml_curl, but they both seemed to have
disappeared. Please can you advise how I can post request so that they can
be viewed as I am stuck on a few things?
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From infos at madovsky.org  Mon Mar 15 17:21:41 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 20:21:41 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
Message-ID: <82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>

Ok, for what reason ?
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 5:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Word of advice.  MP3 is not recommended at all.


  /b


  On Mar 15, 2010, at 3:59 PM, Madovsky wrote:


    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz




------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 17:31:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 19:31:14 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
	<82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
Message-ID: <306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>

You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.

/b

On Mar 15, 2010, at 7:21 PM, Madovsky wrote:

> Ok, for what reason ?
> 

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From infos at madovsky.org  Mon Mar 15 17:46:27 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 20:46:27 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
Message-ID: <EA0AE9A51FD940A7B920337A3DCCE5F1@MOBILEE1705>

ok I understand.
So any way to avoid wav (which could be big for an email) ?
I guess I can use the audio format of modules I compiled in FS isn't it ?

  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 8:31 PM
  Subject: Re: [Freeswitch-users] voicemail options


  You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.


  /b


  On Mar 15, 2010, at 7:21 PM, Madovsky wrote:


    Ok, for what reason ?






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From spencer at 5ninesolutions.com  Mon Mar 15 17:55:56 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Mon, 15 Mar 2010 17:55:56 -0700
Subject: [Freeswitch-users] Radius CDR
Message-ID: <BE35BA86-E45D-4BCD-8AB7-4BD40BE561E5@5ninesolutions.com>

Hello all,
Is mod_radius_cdr still around and does anyone have any experience  
with it?

Spencer




From infos at madovsky.org  Mon Mar 15 19:26:04 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 22:26:04 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
Message-ID: <FB6881D5489143338D823F52AE004230@MOBILEE1705>

now I can see after changed in default config with wav :

/bin/cat: write error: Broken pipe
sh: line 1: 22006 Done(1)                 /bin/cat /tmp/mail.12687062250d3d
     22007 Segmentation fault      | sendmail -f 123456789 at mydomain -t owner at mydomain

however it seems that the file exists in /tmp but in a different subfolder and different name

/tmp/<mydomain>/<userid>/msg_99010fd0-f429-441f-960d-939bb545591c.wav

Amy idea ?


  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 8:31 PM
  Subject: Re: [Freeswitch-users] voicemail options


  You can't expect mp3 to scale.  Also if you happen to hit a bad MP3 that crashes the decoder... which can happen the whole switch goes with it.


  /b


  On Mar 15, 2010, at 7:21 PM, Madovsky wrote:


    Ok, for what reason ?






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dujinfang at gmail.com  Mon Mar 15 20:09:00 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 11:09:00 +0800
Subject: [Freeswitch-users] LUA Call duration or Billing Sec
In-Reply-To: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
References: <8a19bf2e1003150640ld95ba1dqd888a9cedf6df381@mail.gmail.com>
Message-ID: <23f91031003152009o7c67cb66ubdcba84bd41c040c@mail.gmail.com>

Using event consumer immediately after session:hangup

-- uuid = session.uuid
session:hangup
   for e in (function() return con:pop(1) end) do
             -- psudo code
             -- if e.uuid ==  uuid and e.event_type == reporting then
....
   end

note this is not safe since you may miss the event before you do
con:pop, then maybe you can try non-blocking con:pop() ...  not
tested, just an idea. Anyway, doing inline bill is not recommended.

2010/3/15 Aloysius Lloyd <lloyd.aloysius at gmail.com>:
> Hi All,
> I have the following simple Lua script answer a inbound call then dial
> another number and bridge the call.
> Is there any way to get the ?call duration?or billing sec using the
> session:getVariable( ?);
>
>
> -- answer the call
> session:answer();
> -- sleep a second
> session:sleep(1000);
> -- Initiate an outbound call
> obSession = freeswitch.Session("sofia/gateway/voipms/14165555555",session)
> obSession:execute("sched_hangup","+60 alloted_timeout");
> -- bride the call
> ?? ?freeswitch.bridge(session, obSession);
> -- hangup
> session:hangup();
>
> Thank you
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Mon Mar 15 20:14:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 22:14:41 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <FB6881D5489143338D823F52AE004230@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705>
	<306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org>
	<FB6881D5489143338D823F52AE004230@MOBILEE1705>
Message-ID: <47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>

Not too sure who Amy is... but did FreeSWITCH crash or your mailer?

/b

On Mar 15, 2010, at 9:26 PM, Madovsky wrote:

> Amy idea ?

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From infos at madovsky.org  Mon Mar 15 20:42:35 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 23:42:35 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705>
	<47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>
Message-ID: <5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>

fingers ripped...
no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

/bin/cat: write error: Broken pipe
sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
     15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

I think it means that the cat doesn't find the file...
so I checked the /tmp permission and it's ok (chmod 1777)
also if I change in voicemail.conf.xml 
<param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
FS continues to use /tmp.

Latest trunk, fedora 10 64bits
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Not too sure who Amy is... but did FreeSWITCH crash or your mailer?


  /b


  On Mar 15, 2010, at 9:26 PM, Madovsky wrote:


    Amy idea ?




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From brian at freeswitch.org  Mon Mar 15 20:50:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 22:50:00 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705>
	<47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org>
	<5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
Message-ID: <D4693B04-452E-453D-B9AF-71B1EC8F9EA6@freeswitch.org>

NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using?

/b

On Mar 15, 2010, at 10:42 PM, Madovsky wrote:

> fingers ripped...
> no FS is ok, I updated to 16998,  used the default voicemail config and it's the same
>  
> /bin/cat: write error: Broken pipe
> sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
>      15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba
>  
> I think it means that the cat doesn't find the file...
> so I checked the /tmp permission and it's ok (chmod 1777)
> also if I change in voicemail.conf.xml
> <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
> FS continues to use /tmp.
>  
> Latest trunk, fedora 10 64bits
> 

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From infos at madovsky.org  Mon Mar 15 20:58:30 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 15 Mar 2010 23:58:30 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705><2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org><82909F7AEAFD4F579399C90802F1476B@MOBILEE1705><306F88D0-0B49-4772-9451-F9667FC6AA7F@freeswitch.org><FB6881D5489143338D823F52AE004230@MOBILEE1705><47AC86C6-784D-4AD1-A7B5-F6F925B3DC29@freeswitch.org><5EBFD6DBC1194098B33C8092B54C54FB@MOBILEE1705>
	<D4693B04-452E-453D-B9AF-71B1EC8F9EA6@freeswitch.org>
Message-ID: <E6104FD0533A4D1B91F45A2AE37A1ED3@MOBILEE1705>

sendmail.
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:50 PM
  Subject: Re: [Freeswitch-users] voicemail options


  NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using?


  /b


  On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


    fingers ripped...
    no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

    /bin/cat: write error: Broken pipe
    sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
         15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

    I think it means that the cat doesn't find the file...
    so I checked the /tmp permission and it's ok (chmod 1777)
    also if I change in voicemail.conf.xml
    <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
    FS continues to use /tmp.

    Latest trunk, fedora 10 64bits






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 15 21:20:06 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:20:06 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>

So I checked t/mp WHILE the operator says "message saved"
but nothing like /tmp/mail.xxxxxxx

do you have an army of ideas ?
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 11:58 PM
  Subject: Re: [Freeswitch-users] voicemail options


  sendmail.
    ----- Original Message ----- 
    From: Brian West 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Monday, March 15, 2010 11:50 PM
    Subject: Re: [Freeswitch-users] voicemail options


    NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


    /b


    On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


      fingers ripped...
      no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

      /bin/cat: write error: Broken pipe
      sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
           15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

      I think it means that the cat doesn't find the file...
      so I checked the /tmp permission and it's ok (chmod 1777)
      also if I change in voicemail.conf.xml
      <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
      FS continues to use /tmp.

      Latest trunk, fedora 10 64bits






----------------------------------------------------------------------------


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 21:28:34 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:28:34 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>
References: <92F81BC5148C4B688B38497FD5216AD2@MOBILEE1705>
Message-ID: <90B89EA1-AF0B-4A4F-83C1-A5B208D86BE9@freeswitch.org>

Yes your mailer is crashing and never actually mailing the file... Works here for me.... I use exim see faq on wiki.

/b

On Mar 15, 2010, at 11:20 PM, Madovsky wrote:

> So I checked t/mp WHILE the operator says "message saved"
> but nothing like /tmp/mail.xxxxxxx
>  
> do you have an army of ideas ?

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From tayeb.meftah at gmail.com  Mon Mar 15 21:30:44 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:30:44 +0100
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <4B9F0974.70509@gmail.com>

yeah, this is right but svn up to 1.0.5!
you are 3yeers old
Le 15/03/2010 16:36, srinivasula reddy a ?crit :
> Hi,
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco 
> like just farwading the call.
> <action application="bridge" data= 
> <mailto:data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>sofia/internal/${destination_number}@ciscogatewayip 
> <mailto:sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>/>
> i do not using any gateway here, just farwarding call to router, is 
> this correct way of doing? any idea
>
> -- 
> Srinivasula Reddy K
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From tzury.by at reguluslabs.com  Mon Mar 15 21:31:28 2010
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Tue, 16 Mar 2010 06:31:28 +0200
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com> 
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com> 
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com> 
	<c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
Message-ID: <10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>

On Mon, Mar 15, 2010 at 5:28 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Ok, so the call is not even being placed in the ISDN network.
>
> Did you manage to get the sangoma_prid logs working? if that's the case
> then paste them too, it seems sangoma_prid is not seeing the request from
> FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
> response from sangoma_prid binary (they both communicate through sctp
> socket).
>

This is the log (a result of `# /usr/sbin/sangoma_prid &> sangoma_prid.log`)

================System restart=============
= Sangoma PRI Protocol Stack Daemon       =
= Version: 1.58                           =
= Date: Feb 18 2010                       =
= Wanpipe Release: wanpipe-3.5.10.3          =
= Revision:Revision: 15364          =
===========================================
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.overload.task
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.100%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.90%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.overload.task
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.100%-full
DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
system.task.queue.90%-full
WARN - detected an AFT-A101 board with an unsupported firmware version.
(detected: 0x34, supported: 0x36). You may encounter unknown issues if you
use an unsupported firmware version. Please contact support to get the
proper instructions to get the supported firmware unless you have been
instructed to use this firmware version.
ERROR - Assertion failed (in_eNewState == STARTING_TSLINK_STATE)
file=../../../stack/src/main.cpp line=134
ERROR - ACE_Task::wait failed with error 22 (Invalid argument) . Task name =
TelesoftStack
ERROR - failed to waitUntilDone the TaskManager's thread TelesoftStack
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From infos at madovsky.org  Mon Mar 15 21:33:58 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:33:58 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>

there's a thread here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
for exim with the same problem,
but at the end he didn't say if it worked or not
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:20 AM
  Subject: Re: [Freeswitch-users] voicemail options


  So I checked t/mp WHILE the operator says "message saved"
  but nothing like /tmp/mail.xxxxxxx

  do you have an army of ideas ?
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Monday, March 15, 2010 11:58 PM
    Subject: Re: [Freeswitch-users] voicemail options


    sendmail.
      ----- Original Message ----- 
      From: Brian West 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 11:50 PM
      Subject: Re: [Freeswitch-users] voicemail options


      NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


      /b


      On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


        fingers ripped...
        no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

        /bin/cat: write error: Broken pipe
        sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
             15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

        I think it means that the cat doesn't find the file...
        so I checked the /tmp permission and it's ok (chmod 1777)
        also if I change in voicemail.conf.xml
        <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
        FS continues to use /tmp.

        Latest trunk, fedora 10 64bits






--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 15 21:38:21 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:38:21 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>
References: <3B08FAEF58694375A8A2739BEC293BDA@MOBILEE1705>
Message-ID: <6C5774D8-D130-458F-BC40-0DC1F5ECE87B@freeswitch.org>

read the wiki it tells you how.

/b

On Mar 15, 2010, at 11:33 PM, Madovsky wrote:

> there's a thread here
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
> for exim with the same problem,
> but at the end he didn't say if it worked or not
> 

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From brian at freeswitch.org  Mon Mar 15 21:39:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 15 Mar 2010 23:39:44 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <7E79BFF7-FC88-4333-AD25-370D55BB49AF@freeswitch.org>

You should NEVER and I mean NEVER use ${destination_number} as is... its possible to inject things into the dial string.  We never recommend it.

gun+bullets=shot in foot.

/b

On Mar 15, 2010, at 10:36 AM, srinivasula reddy wrote:

> <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>

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From tayeb.meftah at gmail.com  Mon Mar 15 21:44:53 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:44:53 +0100
Subject: [Freeswitch-users] latest windows binary
Message-ID: <4B9F0CC5.3020409@gmail.com>

hi all,
could anyone build a windows version and put it in latest.freeswitch.org?
thanks



From infos at madovsky.org  Mon Mar 15 21:46:41 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:46:41 -0400
Subject: [Freeswitch-users] voicemail options [SOLVED]
Message-ID: <AD69B0257309443D943A3F48754B9C00@MOBILEE1705>

as my sendmail is ultra secure I forgot that the FROM "email"
MUST be registered in your mails server...

but now there is another problem. I open a new thread

Thanks for your time

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:33 AM
  Subject: Re: [Freeswitch-users] voicemail options


  there's a thread here
  http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
  for exim with the same problem,
  but at the end he didn't say if it worked or not
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 12:20 AM
    Subject: Re: [Freeswitch-users] voicemail options


    So I checked t/mp WHILE the operator says "message saved"
    but nothing like /tmp/mail.xxxxxxx

    do you have an army of ideas ?
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, March 15, 2010 11:58 PM
      Subject: Re: [Freeswitch-users] voicemail options


      sendmail.
        ----- Original Message ----- 
        From: Brian West 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Monday, March 15, 2010 11:50 PM
        Subject: Re: [Freeswitch-users] voicemail options


        NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


        /b


        On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


          fingers ripped...
          no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

          /bin/cat: write error: Broken pipe
          sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
               15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

          I think it means that the cat doesn't find the file...
          so I checked the /tmp permission and it's ok (chmod 1777)
          also if I change in voicemail.conf.xml
          <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
          FS continues to use /tmp.

          Latest trunk, fedora 10 64bits






------------------------------------------------------------------------


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org
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From infos at madovsky.org  Mon Mar 15 21:49:33 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:49:33 -0400
Subject: [Freeswitch-users] voicemail message "saved" error
Message-ID: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>

Hi,

2010-03-16 00:44:27.878289 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns
sh: line 1: 30667 Done                    /bin/cat /tmp/mail.126871466794cf
     30668 Segmentation fault      | sendmail -f infos at bibi.baba -t other at kiki.coco


Thanks

F
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From tayeb.meftah at gmail.com  Mon Mar 15 21:55:10 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:55:10 +0100
Subject: [Freeswitch-users] PSTN connection
In-Reply-To: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>
References: <2F2B99BE-23B7-4B44-9157-DFA3791CEF39@ukgb.net>
Message-ID: <4B9F0F2E.4060407@gmail.com>

hi,
mayb you can wait for mod_gsmopen;)
i am not sure.
Le 11/03/2010 18:34, Ken Gillett a ?crit :
> If I am running FreeSwitch on a Mac, would I be able to use Apple's USB modem as a means to connect to the PSTN and utilise this line to place and receive calls?
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    




From tayeb.meftah at gmail.com  Mon Mar 15 21:56:39 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 05:56:39 +0100
Subject: [Freeswitch-users] Attrafax
In-Reply-To: <65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>
References: <4d15ff861003080740t1bd67ee1h73891d2c2155cb5a@mail.gmail.com>	<4B951E63.6060200@coppice.org>	<4d15ff861003080824i1b7d1cb4p1466368d2130f45f@mail.gmail.com>	<A2F1C0E212BBBA4E8C0E7BE568361F0403B383F1FA47@srv02.khdev.corp>	<ADF195FC-6088-4027-A975-62F07BE2A89E@freeswitch.org>	<8b1c9cda1003081439h3da46e03n82ac897d95cf1911@mail.gmail.com>	<E1ACBDDF-ADCD-4D3C-B2CB-7A827F18F5F8@freeswitch.org>	<65d96fc81003100410t7a386f0au411135ae62e34a6b@mail.gmail.com>	<191c3a031003100911x30213038h533d5b89eb364242@mail.gmail.com>
	<65d96fc81003110915v39fb6dbia8e031322f9d451c@mail.gmail.com>
Message-ID: <4B9F0F87.1050001@gmail.com>

mod_fax is ready andd in production no T38 only T30

Le 11/03/2010 18:15, Tihomir Culjaga a ?crit :
>
>
> On Wed, Mar 10, 2010 at 6:11 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     We have a mod_fax for t30 and work on t38 is underway.
>     So we already acknowledge that I guess.
>
>
> anything ready for testing ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From infos at madovsky.org  Mon Mar 15 21:57:55 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 00:57:55 -0400
Subject: [Freeswitch-users] voicemail options [SOLVED]
Message-ID: <F2B4DA51682D4F8E963BF7A657F56FA0@MOBILEE1705>

Brian, I read the wiki 2 days ago already and din't help me
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:46 AM
  Subject: Re: [Freeswitch-users] voicemail options [SOLVED]


  as my sendmail is ultra secure I forgot that the FROM "email"
  MUST be registered in your mails server...

  but now there is another problem. I open a new thread

  Thanks for your time

  Franck
    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 12:33 AM
    Subject: Re: [Freeswitch-users] voicemail options


    there's a thread here
    http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024731.html
    for exim with the same problem,
    but at the end he didn't say if it worked or not
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 16, 2010 12:20 AM
      Subject: Re: [Freeswitch-users] voicemail options


      So I checked t/mp WHILE the operator says "message saved"
      but nothing like /tmp/mail.xxxxxxx

      do you have an army of ideas ?
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Monday, March 15, 2010 11:58 PM
        Subject: Re: [Freeswitch-users] voicemail options


        sendmail.
          ----- Original Message ----- 
          From: Brian West 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, March 15, 2010 11:50 PM
          Subject: Re: [Freeswitch-users] voicemail options


          NO your mailer is crashing cuz its running out of stack space I suspect.  What mailer are you using? 


          /b


          On Mar 15, 2010, at 10:42 PM, Madovsky wrote:


            fingers ripped...
            no FS is ok, I updated to 16998,  used the default voicemail config and it's the same

            /bin/cat: write error: Broken pipe
            sh: line 1: 15281 Done(1)                 /bin/cat /tmp/mail.12687097673f11
                 15282 Segmentation fault      | sendmail -f  gaga at gogo.kiki  -t infos at bobo.baba

            I think it means that the cat doesn't find the file...
            so I checked the /tmp permission and it's ok (chmod 1777)
            also if I change in voicemail.conf.xml
            <param name="storage-dir" value="/tmp"/> to for example <param name="storage-dir" value="/home/tmp"/>
            FS continues to use /tmp.

            Latest trunk, fedora 10 64bits






----------------------------------------------------------------------


          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From tayeb.meftah at gmail.com  Mon Mar 15 22:01:52 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 06:01:52 +0100
Subject: [Freeswitch-users] Switch with External SoftPhone
In-Reply-To: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
References: <eb0385731003101641i1c30d99ew8424e3142bad5b7b@mail.gmail.com>
Message-ID: <4B9F10C0.6020501@gmail.com>

hi
don't confuse betwan internal/external profile
external profile is to send trafic to a provider and to subscrib users
but in internal you can subscrib external and internal users awell
just forward the required ports like 5060TCP/UDP for sip and 16000-32000 
for rtp

Le 11/03/2010 01:41, Vladimir Elizarov a ?crit :
> Hello.
>
> What are the ways connect external users to the fs there? In such a 
> scheme:
> fs with real ip <-> internet <-> nat <-> sip-phone
>
>
> I read the topic http://wiki.freeswitch.org/wiki/External_profile # 
> Switch_with_External_SoftPhone
>
> I did not understand several things:
> - That should be in exteranl5090.xml? Whole profile external.xml c 
> changes in paragraphs 2,3?
> - Sofia/external5090 / <caller extension> @ xxxx: 5090 <- "What goes 
> in xxxx? External UA IP? Domain?" I have the same question. that 
> x.x.x.x should be?
>
> But this seems not quite what I need. Must be external to the 
> subscriber is practically no different from internal.
>
> -- 
> Best regards, Vladimir Elizarov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From brian at freeswitch.org  Mon Mar 15 22:08:22 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 00:08:22 -0500
Subject: [Freeswitch-users] voicemail message "saved" error
In-Reply-To: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>
References: <9E87C048B9AB44A9AE8C7C899740FB21@MOBILEE1705>
Message-ID: <4E102168-434C-4DC7-9D38-661D890BD056@freeswitch.org>

make vm-sync
restart

On Mar 15, 2010, at 11:49 PM, Madovsky wrote:

> 2010-03-16 00:44:27.878289 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_ack]: 'saved' did not match any patterns
> sh: line 1: 30667 Done                    /bin/cat /tmp/mail.126871466794cf
>      30668 Segmentation fault      | sendmail -f infos at bibi.baba -t other at kiki.coco
>  
> Thanks
>  

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From ahmed.ajmal at breezecom.ae  Mon Mar 15 22:41:41 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Tue, 16 Mar 2010 10:41:41 +0500
Subject: [Freeswitch-users] Set/Export channel variable
In-Reply-To: <87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>	<8259813661645760166@unknownmsgid>
	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
Message-ID: <015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>

I am looking at the FS console after hangup is complete. I also have a ESL
event hooked up for HANGUP_COMPLETE event. In both cases I am unable to see
the sip_gateway_name variable.

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 15, 2010 11:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

 

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>
wrote:

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Where are you looking to find the variable?
-MC

 

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From ahmed.ajmal at breezecom.ae  Mon Mar 15 23:21:49 2010
From: ahmed.ajmal at breezecom.ae (Ahmed Ajmal)
Date: Tue, 16 Mar 2010 11:21:49 +0500
Subject: [Freeswitch-users] Set/Export channel variable [SOLVED]
In-Reply-To: <015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>	<8259813661645760166@unknownmsgid>	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
	<015c01cac4cb$5f02baa0$1d082fe0$@ajmal@breezecom.ae>
Message-ID: <016701cac4d0$f917d210$eb477630$@ajmal@breezecom.ae>

So it seems that the FS console by default only displays the 'a' leg vars
which is why I couldn't see it (and it makes sense too since gateway is on
the b leg side). I have the ESL script now display both the inbound/outbound
after which I can see the sip_gateway_name var. 

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ahmed
Ajmal
Sent: Tuesday, March 16, 2010 10:42 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

I am looking at the FS console after hangup is complete. I also have a ESL
event hooked up for HANGUP_COMPLETE event. In both cases I am unable to see
the sip_gateway_name variable.

 

-ahmed

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, March 15, 2010 11:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set/Export channel variable

 

 

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>
wrote:

Hi

 

I have now configured the gateways as suggested below but I am not able to
see the sip_gateway_name variable when the call hangup event is complete. I
am on Freeswitch 1.0.5.  

 

Where are you looking to find the variable?
-MC

 

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From infos at madovsky.org  Mon Mar 15 23:32:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 02:32:10 -0400
Subject: [Freeswitch-users] voicemail options
References: <21B81B3B0E304C759391E0C4E657E59F@MOBILEE1705><87f2f3b91003151123n1f340c4eu1d61bccabe73cc09@mail.gmail.com><9CAB6A7159E24717B31BB21E01FD16E4@MOBILEE1705><c29b930b1003151232x7a3580ccl8030e018f8912762@mail.gmail.com><A6BE7466871D488AA1C12B54487CCAB1@MOBILEE1705><c29b930b1003151310v26fdfa8aod6290b8577d143c4@mail.gmail.com><180F642C4A184B1FAC86489E19B28AC0@MOBILEE1705>
	<2456D54C-A539-4BBA-AF86-937E9BC2F1A1@freeswitch.org>
Message-ID: <1B3CCF4C5CD84CF28FA7270D0C20EA2B@MOBILEE1705>

Sorry Brian, I just realised that
the wav is encoded in 8khz for vociemail.
So it's not  big MB deal

Good Night
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 15, 2010 5:14 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Word of advice.  MP3 is not recommended at all.


  /b


  On Mar 15, 2010, at 3:59 PM, Madovsky wrote:


    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 LAME 3.97 64bits (http://www.mp3dev.org/)
    2010-03-15 16:55:12.846384 [INFO] mod_shout.c:297 Using polyphase lowpass filter, transition band:  7806 Hz -  8000 Hz




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tayeb.meftah at gmail.com  Tue Mar 16 01:00:23 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 16 Mar 2010 09:00:23 +0100
Subject: [Freeswitch-users] answering call in irly_media mode
Message-ID: <4B9F3A97.4030609@gmail.com>

hi all
i want to setup a custom ringback tone for inbound (not outbound calls)
is in irly_media mode but i still heare the normal US ringback.
where is my dialplan extension:
http://pastebin.freeswitch.org/12452
thanks



From will.traenkle at yahoo.com  Tue Mar 16 01:03:32 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 16 Mar 2010 01:03:32 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
	<923176.22123.qm@web57601.mail.re1.yahoo.com>
	<87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
Message-ID: <617776.63524.qm@web57601.mail.re1.yahoo.com>

Michael,

Thanks for your help.  I was able to get it work, however, I would like your opinion as how to do this correctly as I have a hunch I may have opened up some security risks or have gone against freeswitch best practices.  If not, just let me know and I will leave it as is.

I modified the public.xml file under /conf/dialplan/


I uncommented the following:

<!-- TQL
I removed this because of the auto attendant to hunt group to external number error.
I am sure there is some security or other issue in doing this.
But hey, at least my hunt groups work.

    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>
-->

I added the following:

<!-- TQL
I added this to pick up calls being sent to voicemail.
Note, put this after public_extensions
-->

    <extension name="send_to_voicemail">
      <condition field="destination_number" expression="^\*99(\d{3,6})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

<!-- TQL
I added this to pick up the hunt groups. 
Note, put this after the public/*.xml includes
-->

        <X-PRE-PROCESS cmd="include" data="default/*.xml"/>






________________________________
From: Michael Collins <msc at freeswitch.org>
To: William Traenkle <will.traenkle at yahoo.com>
Sent: Mon, March 15, 2010 5:11:57 PM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds




On Mon, Mar 15, 2010 at 4:22 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials
>an internal extension via correctly but when trying to dial an
>external number such a cell phone via sip uri it is unable to connect
>or ring the cell phone and it waits the correct timeout period before it
>goes to voicemail.
>
>However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  
>
>
>
>>There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.
>
>>1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR
>
>>2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3
In #2 your call to 7000 is hitting the public context. Most likely this isn't what you were anticipating. Try handling 7000 in the public context. Find the public_extensions dp entry and update your regex there:
<condition field="destination_number" expression="^(10[01][0-9]|7000)$">

reloadxml and try again. If it works then consider creating a public context extension that specifically handles these calls, perhaps checking the IP address of the source of the call. 
-MC


>
>>3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3
>
>>4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
>
> 
>5) Dialplan default.xml:  
>http://pastebin.com/iAbNSW9E 
> 
>If you require additional information or have any other
>questions, please let me know.
>
>>Thanks,
>
>>-Will
>
________________________________
 From: Michael Collins <msc at freeswitch.org>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tue, March 9, 2010 11:00:53 PM
>Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
>
>
>
>
>On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:
>
>When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>>
>>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>>
>>Any ideas where I should look and what line of code I might be missing?
>>
>>
>
>Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
>>
>-MC
>
>
>



      
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From will.traenkle at yahoo.com  Tue Mar 16 01:09:53 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Tue, 16 Mar 2010 01:09:53 -0700 (PDT)
Subject: [Freeswitch-users] IVR to hunt group to external destination
	fails // direct to hunt group to external desitnation succeeds
Message-ID: <227536.63049.qm@web57605.mail.re1.yahoo.com>

Michael,

Last question, what dictates when a call uses the public.xml or the default.xml dial plan?

Thanks,





________________________________
From: William Traenkle <will.traenkle at yahoo.com>
To: Michael Collins <msc at freeswitch.org>; freeswitch-users at lists.freeswitch.org
Sent: Tue, March 16, 2010 1:03:32 AM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds


Michael,

Thanks for your help.  I was able to get it work, however, I would like your opinion as how to do this correctly as I have a hunch I may have opened up some security risks or have gone against freeswitch best practices.  If not, just let me know and I will leave it as is.

I modified the public.xml file under /conf/dialplan/


I uncommented the following:

<!-- TQL
I removed this because of the auto attendant to hunt group to external number error.
I am sure there is some security or other issue in doing this.
But hey, at least my hunt groups work.

    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>
-->

I added the following:

<!-- TQL
I added this to pick up calls being sent to voicemail.
Note, put this after public_extensions
-->

    <extension name="send_to_voicemail">
      <condition field="destination_number" expression="^\*99(\d{3,6})$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

<!-- TQL
I added this to pick up the hunt groups. 
Note, put this after the public/*.xml includes
-->

        <X-PRE-PROCESS cmd="include" data="default/*.xml"/>






________________________________
From: Michael Collins <msc at freeswitch.org>
To: William Traenkle <will.traenkle at yahoo.com>
Sent: Mon, March 15, 2010 5:11:57 PM
Subject: Re: IVR to hunt group to external destination fails // direct to hunt  group to external desitnation succeeds




On Mon, Mar 15, 2010 at 4:22 PM, William Traenkle <will.traenkle at yahoo.com> wrote:

When a hunt group is accessed from the main IVR, the hunt group dials
>an internal extension via correctly but when trying to dial an
>external number such a cell phone via sip uri it is unable to connect
>or ring the cell phone and it waits the correct timeout period before it
>goes to voicemail.
>
>However, if you dial the hunt group directly without going through the main IVR, the external number such as cell phone that uses sip uri to connect to it work perfect.  
>
>
>
>>There is only an issue when the hunt group is accessed from the main IVR when dialing an external number using sip uri.
>
>>1) fs_cli output for successful call scenario directly to x7000 hunt group: http://pastebin.com/Q1ayXkNR
>
>>2) fs_cli output for failed call scenario x5002 IVR to x7000 hunt group: http://pastebin.com/FaxuGdJ3
In #2 your call to 7000 is hitting the public context. Most likely this isn't what you were anticipating. Try handling 7000 in the public context. Find the public_extensions dp entry and update your regex there:
<condition field="destination_number" expression="^(10[01][0-9]|7000)$">

reloadxml and try again. If it works then consider creating a public context extension that specifically handles these calls, perhaps checking the IP address of the source of the call. 
-MC


>
>>3) autoattendant_5002.js script: http://pastebin.com/FaxuGdJ3
>
>>4) huntgroup_7000.js script: http://pastebin.com/wNX9K7Ab
>
> 
>5) Dialplan default.xml:  
>http://pastebin.com/iAbNSW9E 
> 
>If you require additional information or have any other
>questions, please let me know.
>
>>Thanks,
>
>>-Will
>
________________________________
 From: Michael Collins <msc at freeswitch.org>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tue, March 9, 2010 11:00:53 PM
>Subject: Re: [Freeswitch-users] IVR to hunt group to external destination fails // direct to hunt group to external desitnation succeeds
>
>
>
>
>On Tue, Mar 9, 2010 at 9:55 PM, William Traenkle <will.traenkle at yahoo.com> wrote:
>
>When a hunt group is accessed from the main IVR, the hunt group dials an internal extension via transfer correctly but when trying to dial an external number such a cell phone via sip uri it is unable to connect or ring the phone but it waits the correct timeout period before it goes to voicemail.
>>
>>However, if you access the hunt group directly without going through the main IVR, all works perfectly.  
>>
>>Any ideas where I should look and what line of code I might be missing?
>>
>>
>
>Start by capturing the fs_cli debug output for the IVR call. Look through the debug lines for clues as to what is happening with your call. The symptom "unable to connect" has many possible causes, so try to locate the debug lines leading up to the attempt to bridge the SIP URI. If you want community help then pastebin the full debug log plus the relevant XML: the dialplan extension and the IVR definition.
>>
>-MC
>
>
>


      
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From jason at jasonjgw.net  Tue Mar 16 01:34:58 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Mar 2010 19:34:58 +1100
Subject: [Freeswitch-users] IVR to hunt group to external destination
 fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <227536.63049.qm@web57605.mail.re1.yahoo.com>
References: <227536.63049.qm@web57605.mail.re1.yahoo.com>
Message-ID: <20100316083458.GA30096@jdc.jasonjgw.net>

William Traenkle <will.traenkle at yahoo.com> wrote:
> 
> Last question, what dictates when a call uses the public.xml or the
> default.xml dial plan?

The context of the call, e.g., as specified in the SIP profile.




From jason at jasonjgw.net  Tue Mar 16 01:45:09 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 16 Mar 2010 19:45:09 +1100
Subject: [Freeswitch-users] IVR to hunt group to external destination
 fails // direct to hunt group to external desitnation succeeds
In-Reply-To: <617776.63524.qm@web57601.mail.re1.yahoo.com>
References: <153857.43476.qm@web57613.mail.re1.yahoo.com>
	<87f2f3b91003092300n4bde010dy23b1374931058a@mail.gmail.com>
	<923176.22123.qm@web57601.mail.re1.yahoo.com>
	<87f2f3b91003151711r11746d5bt27e02ce54a6e98e@mail.gmail.com>
	<617776.63524.qm@web57601.mail.re1.yahoo.com>
Message-ID: <20100316084509.GA30197@jdc.jasonjgw.net>

William Traenkle <will.traenkle at yahoo.com> wrote:
> Thanks for your help.  I was able to get it work, however, I would like your
> opinion as how to do this correctly as I have a hunch I may have opened up
> some security risks or have gone against freeswitch best practices.  If not,
> just let me know and I will leave it as is.

What you don't want is for calls originating in the public context to be
transferred into extensions of the default dial plan for which they aren't
authorized.

Based on a cursory reading of your post, I don't notice any problems; you
aren't transferring the calls into the default dial plan using the transfer
application, so it should all just work fine.

Having said that, I'm not responsible for your security; it's your task to
consider call flows and make sure that public calls won't end up where you
don't want them.




From dujinfang at gmail.com  Tue Mar 16 02:53:21 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 16 Mar 2010 17:53:21 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
Message-ID: <23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> FS would prefer you run your kernel at 1000hz you are probably on a VM if
> its 10000hz
>

It's a  physical machine. And I'm also running 20 skype clients. as
100HZ is recommended for skypiax. I think the kernel should be 100 or
250 HZ. weird.


here is the dingaling log with dl_debug on

http://pastebin.freeswitch.org/12453
My public IP is 220.231.26.130, and nat to 192.168.1.27(me)


still get this message on client:

Sorry! The voice chat with language_lesson failed because of a network
problem at 5:39 PM. Please try again.
 Click here to report this to Google



Thanks.









>
> On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> Windows client and web site on Mac. They both work on the other two
>> instances of FreeSWITCH I mentioned in this thread.
>>
>> btw, can you give me a hint on the timer warning?
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > you would have to execute
>> >
>> > dl_debug on
>> >
>> > to get a trace that would tell us anything about that?
>> >
>> > Are you using the windows gtalk client or the web site?
>> > We are following google who have been pushing for the online client
>> > embedded
>> > in the gmail website.
>> >
>> >
>> >
>> >
>> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> here is the client:
>> >>
>> >> <profile type="client">
>> >> ?<param name="name" value="gmail.com"/>
>> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> ?<param name="password" value="password"/>
>> >> ?<param name="server" value="talk.google.com"/>
>> >> ?<param name="dialplan" value="XML"/>
>> >> ?<param name="contex" value="public"/>
>> >> ?<param name="message" value="FreeSWITCH"/>
>> >> ?<param name="rtp-ip" value="auto"/>
>> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> ?<param name="auto-login" value="true"/>
>> >> ?<param name="auto-reply" value="Welcome"/>
>> >> ?<param name="sasl" value="plain"/>
>> >> ?<param name="tls" value="true"/>
>> >> ?<param name="exten" value="5000"/>
>> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> </profile>
>> >>
>> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> > ALL,
>> >> >
>> >> > I just upgrade FS to 16992.
>> >> >
>> >> > 1) timer warning: How's possible the 10000 microseconds? audio sounds
>> >> > ok.
>> >> >
>> >> >
>> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> >> > detected!
>> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> > experience audio problems.
>> >> >
>> >> > uname -a
>> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05 UTC
>> >> > 2008 i686 GNU/Linux
>> >> >
>> >> >
>> >> > I used a script to check the interupts
>> >> >
>> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >
>> >> >
>> >> > 2) run dingaling with client mode. dingaling doesn't pick up answer.
>> >> > I
>> >> > originate a call from FS, the console blocks, gtalk client ring and
>> >> > it
>> >> > looks ok when I hit answer, but no sound, FS console still blocks. a
>> >> > few seconds later gtalk client shows:
>> >> >
>> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> > network
>> >> > problem at 5:50 PM. Please try again.
>> >> > ?Click here to report this to Google.
>> >> >
>> >> >
>> >> > here is a log with bgapi:
>> >> > http://pastebin.freeswitch.org/12434
>> >> >
>> >> >
>> >> > rev 14696 works on the same server same conf. another server running
>> >> > 16958M works at the same time. I have no clue to check, is it related
>> >> > to the timer as in 1)? can some help me to take a look? let me know
>> >> > if
>> >> > you need "dl_debug on" logs.
>> >> >
>> >> > Thanks.
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jbrucehopkins at gmail.com  Tue Mar 16 03:30:43 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 10:30:43 +0000
Subject: [Freeswitch-users] Speex transcoding question
Message-ID: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>

Hi,

I am having trouble getting my configuration  right so that I can have a
call transcoded to Speex wideband from another codec (alaw or g.722).

If both phones use Speex wideband with no transcoding required by FS, the
call succeeds though.

My codecs are listed in vars.xml as follows:
 <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
@20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
@20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

However if I make a call from a phone A using, say, g.722 to a phone using
Speex wideband, the SDP in the invite from phone A to phone B does not
include Speex wideband.  In fact the SDP includes speex/8000 even though
speex/8000 is neither enabled in vars.xml, not in either of the phones.
Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
B (from Wireshark) :

Media Attribute (a): rtpmap:9 G722/8000
Media Attribute (a): rtpmap:99 SPEEX/8000
Media Attribute (a): rtpmap:115 G7221/32000
Media Attribute (a): fmtp:115 bitrate=48000
Media Attribute (a): rtpmap:107 G7221/16000
Media Attribute (a): fmtp:107 bitrate=32000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): rtpmap:13 CN/8000
Media Attribute (a): ptime:20

Could anyone tell me what I am doing wrong please?

Many thanks
Bruce
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From jbrucehopkins at gmail.com  Tue Mar 16 03:36:01 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 10:36:01 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
Message-ID: <d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>

Please ignore this last email.  I think I have missed out something I hav
already been told.  Sorry for the waste of bandwidth.

regards
Bruce

On 16 March 2010 10:30, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Hi,
>
> I am having trouble getting my configuration  right so that I can have a
> call transcoded to Speex wideband from another codec (alaw or g.722).
>
> If both phones use Speex wideband with no transcoding required by FS, the
> call succeeds though.
>
> My codecs are listed in vars.xml as follows:
>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>
> However if I make a call from a phone A using, say, g.722 to a phone using
> Speex wideband, the SDP in the invite from phone A to phone B does not
> include Speex wideband.  In fact the SDP includes speex/8000 even though
> speex/8000 is neither enabled in vars.xml, not in either of the phones.
> Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
> B (from Wireshark) :
>
> Media Attribute (a): rtpmap:9 G722/8000
> Media Attribute (a): rtpmap:99 SPEEX/8000
> Media Attribute (a): rtpmap:115 G7221/32000
> Media Attribute (a): fmtp:115 bitrate=48000
> Media Attribute (a): rtpmap:107 G7221/16000
> Media Attribute (a): fmtp:107 bitrate=32000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:3 GSM/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
> Media Attribute (a): rtpmap:13 CN/8000
> Media Attribute (a): ptime:20
>
> Could anyone tell me what I am doing wrong please?
>
> Many thanks
> Bruce
>
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From ken at ukgb.net  Tue Mar 16 03:50:05 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 10:50:05 +0000
Subject: [Freeswitch-users] call info display
Message-ID: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>

Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.

When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).

Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?

So, how does FS handle all the above?
What does it pass to the recipient, for both external and internal calls?
Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?


Ken G i l l e t t

_/_/_/_/_/_/_/_/




From siniypin at gmail.com  Tue Mar 16 04:18:55 2010
From: siniypin at gmail.com (RobertT)
Date: Tue, 16 Mar 2010 14:18:55 +0300
Subject: [Freeswitch-users] ICE related sdp headers forwarding
In-Reply-To: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>
References: <2160023e1003090117w159d72fdsf1914c52460aaf98@mail.gmail.com>
Message-ID: <2160023e1003160418m163f1badq4fea3df25456fe04@mail.gmail.com>

Should someone want to know the answers I found, here they are.

I had two ideas about how to forward SDP ICE-related headers from FS to
leg-B.
*1. Redirect a directly to b. But to accomplish this I've to know b's IP,
that it was registered with, which is being stored in sip_registrations
db. Can I obtain it with db app in dialplan and substitute as a parameter to
deflect app?
*This is* *rather simple - just call "defend" app from dialplan with
${destination_number} parameter. Actually, FS answers with a bit awkward
address in Refer-to header, containing <profile_name><destination>@<domain>,
so one would have to parse recieved address manually, to use a correct sip
uri. Did I make something wrong?..
*
*
*2. Force FS to copy remote SDP headers, or at least a part of them, to it's
local SDP.*
This is even more simple - just set inbound_bypass_media to true in sofia
profile settings, and FS will leave all of the SDP headers intact. But, you
have to remember, that media traffic for this profile will bypass FS, as
parameter's name implies. Therefore FS will not be able to manipulate media
data in anyway.

Best regards,
RobertT.
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From jbrucehopkins at gmail.com  Tue Mar 16 04:26:21 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 11:26:21 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
Message-ID: <d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>

OK - sorry to make such a meal of this.

I am still encountering a problem getting my configuration right, having
followed the advice I received previously.

What I am observing is that if more than one flavour of SPEEX is enabled in
vars.xml, then only the first one in the list is offered to phone B in the
SDP section of the INVITE if transcoding is required by Freeswitch.

i.e. In vars.xml I have:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=SPEEX at 16000h,SPEEX at 32000h
,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

 <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=SPEEX at 16000h
,SPEEX at 32000h,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>


If I then try to make a call from Phone A (using PCMA) to Phone B (using
SPEEX/8000), the call fails and the log shows an incompatible destination.
The SDP of the INVITE from Freeswitch to Phone B offers SPEEX/16000 but not
SPEEX/8000 or SPEEX/32000, so there is no compatible codec offered to Phone
B.

If however both Phone A and Phone B both use SPEEX/8000, the call works
fine  - as my default configuration is that the codec preferred by Phone A
goes to the top of the list of codecs offered to Phone B

Could anyone help with what I need to do to resolve my problem when
transcoding is required though please?

I am did "make current" last night so am using the latest snapshop.

many thanks again,
Bruce





On 16 March 2010 10:36, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Please ignore this last email.  I think I have missed out something I hav
> already been told.  Sorry for the waste of bandwidth.
>
> regards
> Bruce
>
>
> On 16 March 2010 10:30, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi,
>>
>> I am having trouble getting my configuration  right so that I can have a
>> call transcoded to Speex wideband from another codec (alaw or g.722).
>>
>> If both phones use Speex wideband with no transcoding required by FS, the
>> call succeeds though.
>>
>> My codecs are listed in vars.xml as follows:
>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> However if I make a call from a phone A using, say, g.722 to a phone using
>> Speex wideband, the SDP in the invite from phone A to phone B does not
>> include Speex wideband.  In fact the SDP includes speex/8000 even though
>> speex/8000 is neither enabled in vars.xml, not in either of the phones.
>> Here is the codec listing in  the SDP of the INVITE from Freeswitch to phone
>> B (from Wireshark) :
>>
>> Media Attribute (a): rtpmap:9 G722/8000
>> Media Attribute (a): rtpmap:99 SPEEX/8000
>> Media Attribute (a): rtpmap:115 G7221/32000
>> Media Attribute (a): fmtp:115 bitrate=48000
>> Media Attribute (a): rtpmap:107 G7221/16000
>> Media Attribute (a): fmtp:107 bitrate=32000
>> Media Attribute (a): rtpmap:8 PCMA/8000
>> Media Attribute (a): rtpmap:0 PCMU/8000
>> Media Attribute (a): rtpmap:3 GSM/8000
>> Media Attribute (a): rtpmap:101 telephone-event/8000
>> Media Attribute (a): fmtp:101 0-16
>> Media Attribute (a): rtpmap:13 CN/8000
>> Media Attribute (a): ptime:20
>>
>> Could anyone tell me what I am doing wrong please?
>>
>> Many thanks
>> Bruce
>>
>
>
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From peter.olsson at visionutveckling.se  Tue Mar 16 04:41:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 12:41:09 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>

I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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From Prometheus001 at gmx.net  Tue Mar 16 04:55:25 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 16 Mar 2010 12:55:25 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
Message-ID: <4B9F71AD.7070008@gmx.net>

I have the folowing scenario:
  A Snom phone has activated DND (do not disturb)
  When the phone is called, it returns "480 Do Not Disturb" and then
"404 Not Found".
  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
  The PSTN then (wrongly) announces "this number is not listed".

My question: How can we overcome this and announce back a different
message to PSTN, but only on NO_USER_RESPONSE.
I tried to continue on bridge and then this after the bridge command the
following:
      <condition field="variable_originate_disposition"
expression="NO_USER_RESPONSE" inline="true">
        <action application="hangup" data="USER_BUSY"/>
      </condition>

But the installed Freeswith does not support the "inline" yet and we
will not be able to update Freeswitch soon.

So my question:
Is there any other way to achieve this?

Best regards
Peter




From vetali100 at gmail.com  Tue Mar 16 05:04:21 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 14:04:21 +0200
Subject: [Freeswitch-users]  Load balancing solution using pure FreeSWITCH
Message-ID: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>

Hi dear FS community,

After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109)
and (
http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
use MySQL).

It works on few of our servers right now without any issues.

Feel free to reuse / improve it or provide your comments if you see
something wrong with it.


We have one routing server (RS) and several Media Servers (MS).
FreeSWITCH is installed on each server in default configuration.

--> MS_1
RS  --> MS_2
--> MS_N


(Failover can be implemented by adding one more routing server, but this is
out of scope of current description).

Routing server routes calls to media servers based on logic that you can
easily define.
(Generally speaking, Routing server can route part of the calls onto itself,
but to simplify let's say that it will route only to Media Servers).


Routing server is used to register subscribers (it will have directory files
./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
Media servers do NOT contain files with users (security issue? probably
should be managed on dialplan stage of media servers by comparing ip
addresses of requests).


The whole routing can be presented by the following chain:

Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
"sofia/external/" to determined MS IP address-->[Bridge to fail over media
server, optionally]-->End

Let's say we need to route all international calls based on less loaded
server (other logic can be easily implemented in PHP script as well).

*1. RS Dialplan: default.xml*
<include>
        <extension name="route">
                <condition field="destination_number"
expression="^(00)(\d+)">
                        <action application="lua" data="route.lua $2"/>
                </condition>
        </extension>
</include>


*2. RS Lua script: route.lua*
called_number = argv[1]; //without "00"
api = freeswitch.API();
server_ip_address = api:executeString("curl
http://localhost/get_server_ip.php);
forwarding_session_string =
"sofia/external/"..called_number.."@"..server_ip_address;
session:setVariable("bypass_media", "true"); //described below!!!
session:execute("bridge",forwarding_session_string);

*3. PHP script: get_server_ip.php*
<<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
LOADED AT THE MOMENT SERVER>>>
die($less_loaded_server_ip);


All Media Servers should provide information about their current load into
one common database (dedicated server or routing server can be used for
database).
If you use scripts you can update this table on starting a call and after
call finished.
Or you can create a job that will query FreeSWITCH and will update the
database every, say, 10 seconds.


Basically that's all (really!)
I beleive it can be used for most general purpose scenarios without need of
third-party tools.

P.S.
The only problem we had from the beginning was that RTP media traffic was
passing thru both servers: RS and Media Server.
But after we added the following command to the routing script, RTP media
traffic started to pass only thru Media server, making RS almost free.
session:setVariable("bypass_media", "true");



Best regards,
vIT
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From leon at scarlet-internet.nl  Tue Mar 16 05:42:56 2010
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 16 Mar 2010 13:42:56 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <4B9F71AD.7070008@gmx.net>
References: <4B9F71AD.7070008@gmx.net>
Message-ID: <2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>

Hi,

Don't know if it helps, but the inline attribute can only be set on  
actions, not on conditions.

Also, the hangup application is not allowed to be run inline:
http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_dptools/mod_dptools.c?r=HEAD#l3047
(it has no SAF_ROUTING_EXEC flag)

Perhaps you can try to transfer to another context directly after your  
bridge ? (where that second context contains an extension with  
conditions matching on variable_originate_disposition)

kind regards,

Leon


On Mar 16, 2010, at 12:55 PM, Peter P GMX wrote:

> I have the folowing scenario:
>  A Snom phone has activated DND (do not disturb)
>  When the phone is called, it returns "480 Do Not Disturb" and then
> "404 Not Found".
>  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
>  The PSTN then (wrongly) announces "this number is not listed".
>
> My question: How can we overcome this and announce back a different
> message to PSTN, but only on NO_USER_RESPONSE.
> I tried to continue on bridge and then this after the bridge command  
> the
> following:
>      <condition field="variable_originate_disposition"
> expression="NO_USER_RESPONSE" inline="true">
>        <action application="hangup" data="USER_BUSY"/>
>      </condition>
>
> But the installed Freeswith does not support the "inline" yet and we
> will not be able to update Freeswitch soon.
>
> So my question:
> Is there any other way to achieve this?
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tculjaga at gmail.com  Tue Mar 16 06:07:39 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 14:07:39 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
Message-ID: <65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>

this is something you took from Mera .. isn't it ? :)



On Tue, Mar 16, 2010 at 1:04 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi dear FS community,
>
> After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109) and (
> http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
> solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
> use MySQL).
>
> It works on few of our servers right now without any issues.
>
> Feel free to reuse / improve it or provide your comments if you see
> something wrong with it.
>
>
> We have one routing server (RS) and several Media Servers (MS).
> FreeSWITCH is installed on each server in default configuration.
>
> --> MS_1
> RS  --> MS_2
> --> MS_N
>
>
> (Failover can be implemented by adding one more routing server, but this is
> out of scope of current description).
>

routing server is your single point of failure, you should make two of them
in active/standby mode. What about registered endpoints ? ... where do you
register them?.... you should have a DB replication in order to have the
"registered context" distributed between both routing servers


So, i'd suggest to bring up a LAN_HA and have a floating address between
your routing servers.... of course the you should be using ODBC in the core
to have registrations on both servers.




>
> Routing server routes calls to media servers based on logic that you can
> easily define.
> (Generally speaking, Routing server can route part of the calls onto
> itself, but to simplify let's say that it will route only to Media Servers).
>
> you will be loosing calls on the on the MS if it goes down ... maybe there
is a way to "replace" media resource for that calls and keep them going even
after the MS went down.



>
> Routing server is used to register subscribers (it will have directory
> files
> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> Media servers do NOT contain files with users (security issue? probably
> should be managed on dialplan stage of media servers by comparing ip
> addresses of requests).
>
>
consider ODBC in the core!


>
> The whole routing can be presented by the following chain:
>
> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> "sofia/external/" to determined MS IP address-->[Bridge to fail over media
> server, optionally]-->End
>
> Let's say we need to route all international calls based on less loaded
> server (other logic can be easily implemented in PHP script as well).
>
> *1. RS Dialplan: default.xml*
> <include>
>         <extension name="route">
>                 <condition field="destination_number"
> expression="^(00)(\d+)">
>                         <action application="lua" data="route.lua $2"/>
>                 </condition>
>         </extension>
> </include>
>
>
> *2. RS Lua script: route.lua*
> called_number = argv[1]; //without "00"
> api = freeswitch.API();
> server_ip_address = api:executeString("curl
> http://localhost/get_server_ip.php);
> forwarding_session_string =
> "sofia/external/"..called_number.."@"..server_ip_address;
> session:setVariable("bypass_media", "true"); //described below!!!
> session:execute("bridge",forwarding_session_string);
>
>
do your bridging in the dialplan not from a script... you will get better
performances!




> *3. PHP script: get_server_ip.php*
> <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> LOADED AT THE MOMENT SERVER>>>
> die($less_loaded_server_ip);
>
>
> All Media Servers should provide information about their current load into
> one common database (dedicated server or routing server can be used for
> database).
> If you use scripts you can update this table on starting a call and after
> call finished.
> Or you can create a job that will query FreeSWITCH and will update the
> database every, say, 10 seconds.
>
>
> Basically that's all (really!)
> I beleive it can be used for most general purpose scenarios without need of
> third-party tools.
>
> P.S.
> The only problem we had from the beginning was that RTP media traffic was
> passing thru both servers: RS and Media Server.
> But after we added the following command to the routing script, RTP media
> traffic started to pass only thru Media server, making RS almost free.
> session:setVariable("bypass_media", "true");
>
>
>
anyhow it is a good start ... with a custom module for media server
"kepalive" and resource migration it could be a great solution.


T.
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From lawwton at gmail.com  Tue Mar 16 06:13:14 2010
From: lawwton at gmail.com (Alfredo Quiroga-Villamil)
Date: Tue, 16 Mar 2010 09:13:14 -0400
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
Message-ID: <5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>

Vit:

Nice email, very informative; curious to find out the reasons why you
decided to go with FS instead of OpenSip. We soon have to make a
similar decision and would like to find out the rationale behind it.

Thanks,

Alfredo

On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:
> Hi dear FS community,
> After a brief analysis of 2 solutions using OpenSIPS
> and?Kamailio?(http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
> ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my
> own solution using ONLY FreeSWITCH and ONLY 1 table of any database server
> (we use MySQL).
> It works on few of our servers right now without any issues.
> Feel free to reuse / improve it or provide your comments if you see
> something wrong with it.
>
> We have one routing server (RS) and several Media Servers (MS).
> FreeSWITCH is installed on each server in default configuration.
> --> MS_1
> RS ?--> MS_2
> --> MS_N
>
> (Failover can be implemented by adding one more routing server, but this is
> out of scope of current description).
> Routing server routes calls to media servers based on logic that you can
> easily define.
> (Generally speaking, Routing server can route part of the calls onto itself,
> but to simplify let's say that it will route only to Media Servers).
>
> Routing server is used to register subscribers (it will have directory files
> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> Media servers do NOT contain files with users (security issue? probably
> should be managed on dialplan stage of media servers by comparing ip
> addresses of requests).
>
> The whole routing can be presented by the following chain:
> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> "sofia/external/"?to determined MS IP address-->[Bridge to fail over media
> server, optionally]-->End
> Let's say we need to route all international calls based on less loaded
> server (other logic can be easily implemented in PHP script as well).
> 1. RS Dialplan: default.xml
> <include>
> ?? ? ? ?<extension name="route">
> ?? ? ? ? ? ? ? ?<condition field="destination_number"
> expression="^(00)(\d+)">
> ?? ? ? ? ? ? ? ? ? ? ? ?<action application="lua" data="route.lua $2"/>
> ?? ? ? ? ? ? ? ?</condition>
> ?? ? ? ?</extension>
> </include>
>
> 2. RS Lua script: route.lua
> called_number = argv[1]; //without "00"
> api = freeswitch.API();
> server_ip_address = api:executeString("curl
> http://localhost/get_server_ip.php);
> forwarding_session_string =
> "sofia/external/"..called_number.."@"..server_ip_address;
> session:setVariable("bypass_media", "true"); //described below!!!
> session:execute("bridge",forwarding_session_string);
> 3. PHP script: get_server_ip.php
> <<< ?ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> LOADED AT THE MOMENT SERVER>>>
> die($less_loaded_server_ip);
>
> All Media Servers should provide information about their current load into
> one common database (dedicated server or routing server can be used for
> database).
> If you use scripts you can update this table on starting a call and after
> call finished.
> Or you can create a job that will query FreeSWITCH and will update the
> database every, say, 10 seconds.
>
> Basically that's all (really!)
> I beleive it can be used for most general purpose scenarios without need of
> third-party tools.
> P.S.
> The only problem we had from the beginning was that RTP media traffic was
> passing thru both servers: RS and Media Server.
> But after we added the following command to the routing script, RTP media
> traffic started to pass only thru Media server, making RS almost free.
> session:setVariable("bypass_media", "true");
>
>
> Best regards,
> vIT
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From vetali100 at gmail.com  Tue Mar 16 06:19:26 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:19:26 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
Message-ID: <a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>

Hi Tihomir,
What do you mean by "took from Mera"?
If someone else already implemented this, could you please provide me a
link?

This solution was invented by me personally, after looking into the
following: http://wiki.freeswitch.org/wiki/SBC_Setup
They use Kamailio layer as routing, I decided that we can avoid this and use
OOB things.

Thanks for your reply, it can add more value to the solution.

Regards,
vIT



2010/3/16 Tihomir Culjaga <tculjaga at gmail.com>

> this is something you took from Mera .. isn't it ? :)
>
>
>
> On Tue, Mar 16, 2010 at 1:04 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi dear FS community,
>>
>> After a brief analysis of 2 solutions using OpenSIPS and Kamailio (
>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109) and (
>> http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement my own
>> solution using ONLY FreeSWITCH and ONLY 1 table of any database server (we
>> use MySQL).
>>
>> It works on few of our servers right now without any issues.
>>
>> Feel free to reuse / improve it or provide your comments if you see
>> something wrong with it.
>>
>>
>> We have one routing server (RS) and several Media Servers (MS).
>> FreeSWITCH is installed on each server in default configuration.
>>
>> --> MS_1
>> RS  --> MS_2
>> --> MS_N
>>
>>
>> (Failover can be implemented by adding one more routing server, but this
>> is out of scope of current description).
>>
>
> routing server is your single point of failure, you should make two of them
> in active/standby mode. What about registered endpoints ? ... where do you
> register them?.... you should have a DB replication in order to have the
> "registered context" distributed between both routing servers
>
>
> So, i'd suggest to bring up a LAN_HA and have a floating address between
> your routing servers.... of course the you should be using ODBC in the core
> to have registrations on both servers.
>
>
>
>
>>
>> Routing server routes calls to media servers based on logic that you can
>> easily define.
>> (Generally speaking, Routing server can route part of the calls onto
>> itself, but to simplify let's say that it will route only to Media Servers).
>>
>> you will be loosing calls on the on the MS if it goes down ... maybe there
> is a way to "replace" media resource for that calls and keep them going even
> after the MS went down.
>
>
>
>>
>> Routing server is used to register subscribers (it will have directory
>> files
>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>> Media servers do NOT contain files with users (security issue? probably
>> should be managed on dialplan stage of media servers by comparing ip
>> addresses of requests).
>>
>>
> consider ODBC in the core!
>
>
>>
>> The whole routing can be presented by the following chain:
>>
>> Call arrives into Default context-->RS default dialplan-->Lua script-->PHP
>> script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>> "sofia/external/" to determined MS IP address-->[Bridge to fail over media
>> server, optionally]-->End
>>
>> Let's say we need to route all international calls based on less loaded
>> server (other logic can be easily implemented in PHP script as well).
>>
>> *1. RS Dialplan: default.xml*
>> <include>
>>         <extension name="route">
>>                 <condition field="destination_number"
>> expression="^(00)(\d+)">
>>                         <action application="lua" data="route.lua $2"/>
>>                 </condition>
>>         </extension>
>> </include>
>>
>>
>> *2. RS Lua script: route.lua*
>> called_number = argv[1]; //without "00"
>> api = freeswitch.API();
>> server_ip_address = api:executeString("curl
>> http://localhost/get_server_ip.php);
>> forwarding_session_string =
>> "sofia/external/"..called_number.."@"..server_ip_address;
>> session:setVariable("bypass_media", "true"); //described below!!!
>> session:execute("bridge",forwarding_session_string);
>>
>>
> do your bridging in the dialplan not from a script... you will get better
> performances!
>
>
>
>
>> *3. PHP script: get_server_ip.php*
>> <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
>> LOADED AT THE MOMENT SERVER>>>
>> die($less_loaded_server_ip);
>>
>>
>> All Media Servers should provide information about their current load into
>> one common database (dedicated server or routing server can be used for
>> database).
>> If you use scripts you can update this table on starting a call and after
>> call finished.
>> Or you can create a job that will query FreeSWITCH and will update the
>> database every, say, 10 seconds.
>>
>>
>> Basically that's all (really!)
>> I beleive it can be used for most general purpose scenarios without need
>> of third-party tools.
>>
>> P.S.
>> The only problem we had from the beginning was that RTP media traffic was
>> passing thru both servers: RS and Media Server.
>> But after we added the following command to the routing script, RTP media
>> traffic started to pass only thru Media server, making RS almost free.
>> session:setVariable("bypass_media", "true");
>>
>>
>>
> anyhow it is a good start ... with a custom module for media server
> "kepalive" and resource migration it could be a great solution.
>
>
> T.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Tue Mar 16 06:21:58 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:21:58 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
Message-ID: <a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>

Basically this solution gives me exactly what I need at the moment.

If there is a still special reason to use OpenSIPS, maybe we should expect
opinion of some better FS experts then me :-)

vIT

2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>

> Vit:
>
> Nice email, very informative; curious to find out the reasons why you
> decided to go with FS instead of OpenSip. We soon have to make a
> similar decision and would like to find out the rationale behind it.
>
> Thanks,
>
> Alfredo
>
> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
> wrote:
> > Hi dear FS community,
> > After a brief analysis of 2 solutions using OpenSIPS
> > and Kamailio (
> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to implement
> my
> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
> server
> > (we use MySQL).
> > It works on few of our servers right now without any issues.
> > Feel free to reuse / improve it or provide your comments if you see
> > something wrong with it.
> >
> > We have one routing server (RS) and several Media Servers (MS).
> > FreeSWITCH is installed on each server in default configuration.
> > --> MS_1
> > RS  --> MS_2
> > --> MS_N
> >
> > (Failover can be implemented by adding one more routing server, but this
> is
> > out of scope of current description).
> > Routing server routes calls to media servers based on logic that you can
> > easily define.
> > (Generally speaking, Routing server can route part of the calls onto
> itself,
> > but to simplify let's say that it will route only to Media Servers).
> >
> > Routing server is used to register subscribers (it will have directory
> files
> > ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
> > Media servers do NOT contain files with users (security issue? probably
> > should be managed on dialplan stage of media servers by comparing ip
> > addresses of requests).
> >
> > The whole routing can be presented by the following chain:
> > Call arrives into Default context-->RS default dialplan-->Lua
> script-->PHP
> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
> media
> > server, optionally]-->End
> > Let's say we need to route all international calls based on less loaded
> > server (other logic can be easily implemented in PHP script as well).
> > 1. RS Dialplan: default.xml
> > <include>
> >         <extension name="route">
> >                 <condition field="destination_number"
> > expression="^(00)(\d+)">
> >                         <action application="lua" data="route.lua $2"/>
> >                 </condition>
> >         </extension>
> > </include>
> >
> > 2. RS Lua script: route.lua
> > called_number = argv[1]; //without "00"
> > api = freeswitch.API();
> > server_ip_address = api:executeString("curl
> > http://localhost/get_server_ip.php);
> > forwarding_session_string =
> > "sofia/external/"..called_number.."@"..server_ip_address;
> > session:setVariable("bypass_media", "true"); //described below!!!
> > session:execute("bridge",forwarding_session_string);
> > 3. PHP script: get_server_ip.php
> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE LESS
> > LOADED AT THE MOMENT SERVER>>>
> > die($less_loaded_server_ip);
> >
> > All Media Servers should provide information about their current load
> into
> > one common database (dedicated server or routing server can be used for
> > database).
> > If you use scripts you can update this table on starting a call and after
> > call finished.
> > Or you can create a job that will query FreeSWITCH and will update the
> > database every, say, 10 seconds.
> >
> > Basically that's all (really!)
> > I beleive it can be used for most general purpose scenarios without need
> of
> > third-party tools.
> > P.S.
> > The only problem we had from the beginning was that RTP media traffic was
> > passing thru both servers: RS and Media Server.
> > But after we added the following command to the routing script, RTP media
> > traffic started to pass only thru Media server, making RS almost free.
> > session:setVariable("bypass_media", "true");
> >
> >
> > Best regards,
> > vIT
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vetali100 at gmail.com  Tue Mar 16 06:31:13 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 15:31:13 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
Message-ID: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>

Tihomir,
           Could you please add more details to the following?

           "So, i'd suggest to bring up a LAN_HA and have a floating address
between your routing servers.... of course the you should be using ODBC in
the core to have registrations on both servers."

Do you have any links that describe how to implement this kind of fail over?

I thought about configuring 2 DNS servers on 2 Routing servers, so if one
will go down, traffic will be addressed to another, but at the moment I
don't have enough knowledge how to implement this.

Thanks,
vIT



2010/3/16 Vitalii Colosov <vetali100 at gmail.com>

> Basically this solution gives me exactly what I need at the moment.
>
> If there is a still special reason to use OpenSIPS, maybe we should expect
> opinion of some better FS experts then me :-)
>
> vIT
>
> 2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>
>
> Vit:
>>
>> Nice email, very informative; curious to find out the reasons why you
>> decided to go with FS instead of OpenSip. We soon have to make a
>> similar decision and would like to find out the rationale behind it.
>>
>> Thanks,
>>
>> Alfredo
>>
>> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
>> wrote:
>> > Hi dear FS community,
>> > After a brief analysis of 2 solutions using OpenSIPS
>> > and Kamailio (
>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
>> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to
>> implement my
>> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
>> server
>> > (we use MySQL).
>> > It works on few of our servers right now without any issues.
>> > Feel free to reuse / improve it or provide your comments if you see
>> > something wrong with it.
>> >
>> > We have one routing server (RS) and several Media Servers (MS).
>> > FreeSWITCH is installed on each server in default configuration.
>> > --> MS_1
>> > RS  --> MS_2
>> > --> MS_N
>> >
>> > (Failover can be implemented by adding one more routing server, but this
>> is
>> > out of scope of current description).
>> > Routing server routes calls to media servers based on logic that you can
>> > easily define.
>> > (Generally speaking, Routing server can route part of the calls onto
>> itself,
>> > but to simplify let's say that it will route only to Media Servers).
>> >
>> > Routing server is used to register subscribers (it will have directory
>> files
>> >
>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>> > Media servers do NOT contain files with users (security issue? probably
>> > should be managed on dialplan stage of media servers by comparing ip
>> > addresses of requests).
>> >
>> > The whole routing can be presented by the following chain:
>> > Call arrives into Default context-->RS default dialplan-->Lua
>> script-->PHP
>> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
>> media
>> > server, optionally]-->End
>> > Let's say we need to route all international calls based on less loaded
>> > server (other logic can be easily implemented in PHP script as well).
>> > 1. RS Dialplan: default.xml
>> > <include>
>> >         <extension name="route">
>> >                 <condition field="destination_number"
>> > expression="^(00)(\d+)">
>> >                         <action application="lua" data="route.lua $2"/>
>> >                 </condition>
>> >         </extension>
>> > </include>
>> >
>> > 2. RS Lua script: route.lua
>> > called_number = argv[1]; //without "00"
>> > api = freeswitch.API();
>> > server_ip_address = api:executeString("curl
>> > http://localhost/get_server_ip.php);
>> > forwarding_session_string =
>> > "sofia/external/"..called_number.."@"..server_ip_address;
>> > session:setVariable("bypass_media", "true"); //described below!!!
>> > session:execute("bridge",forwarding_session_string);
>> > 3. PHP script: get_server_ip.php
>> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE
>> LESS
>> > LOADED AT THE MOMENT SERVER>>>
>> > die($less_loaded_server_ip);
>> >
>> > All Media Servers should provide information about their current load
>> into
>> > one common database (dedicated server or routing server can be used for
>> > database).
>> > If you use scripts you can update this table on starting a call and
>> after
>> > call finished.
>> > Or you can create a job that will query FreeSWITCH and will update the
>> > database every, say, 10 seconds.
>> >
>> > Basically that's all (really!)
>> > I beleive it can be used for most general purpose scenarios without need
>> of
>> > third-party tools.
>> > P.S.
>> > The only problem we had from the beginning was that RTP media traffic
>> was
>> > passing thru both servers: RS and Media Server.
>> > But after we added the following command to the routing script, RTP
>> media
>> > traffic started to pass only thru Media server, making RS almost free.
>> > session:setVariable("bypass_media", "true");
>> >
>> >
>> > Best regards,
>> > vIT
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mustafa.pk at gmail.com  Tue Mar 16 06:41:05 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Tue, 16 Mar 2010 18:41:05 +0500
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
Message-ID: <8213d6071003160641k4d10c0dcs47c1a94dc05fa080@mail.gmail.com>

On Tue, Mar 16, 2010 at 6:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Tihomir,
>            Could you please add more details to the following?
>
>            "So, i'd suggest to bring up a LAN_HA and have a floating
> address between your routing servers.... of course the you should be using
> ODBC in the core to have registrations on both servers."
>

no problem, you can update your knowledge base for free, check
http://www.linux-ha.org/wiki/Main_Page

>
> Do you have any links that describe how to implement this kind of fail
> over?
>
> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
> will go down, traffic will be addressed to another, but at the moment I
> don't have enough knowledge how to implement this.
>
> Thanks,
> vIT
>
>
>
> 2010/3/16 Vitalii Colosov <vetali100 at gmail.com>
>
> Basically this solution gives me exactly what I need at the moment.
>>
>> If there is a still special reason to use OpenSIPS, maybe we should expect
>> opinion of some better FS experts then me :-)
>>
>> vIT
>>
>> 2010/3/16 Alfredo Quiroga-Villamil <lawwton at gmail.com>
>>
>> Vit:
>>>
>>> Nice email, very informative; curious to find out the reasons why you
>>> decided to go with FS instead of OpenSip. We soon have to make a
>>> similar decision and would like to find out the rationale behind it.
>>>
>>> Thanks,
>>>
>>> Alfredo
>>>
>>> On Tue, Mar 16, 2010 at 8:04 AM, Vitalii Colosov <vetali100 at gmail.com>
>>> wrote:
>>> > Hi dear FS community,
>>> > After a brief analysis of 2 solutions using OpenSIPS
>>> > and Kamailio (
>>> http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id226109
>>> > ) and (http://wiki.freeswitch.org/wiki/SBC_Setup) I decided to
>>> implement my
>>> > own solution using ONLY FreeSWITCH and ONLY 1 table of any database
>>> server
>>> > (we use MySQL).
>>> > It works on few of our servers right now without any issues.
>>> > Feel free to reuse / improve it or provide your comments if you see
>>> > something wrong with it.
>>> >
>>> > We have one routing server (RS) and several Media Servers (MS).
>>> > FreeSWITCH is installed on each server in default configuration.
>>> > --> MS_1
>>> > RS  --> MS_2
>>> > --> MS_N
>>> >
>>> > (Failover can be implemented by adding one more routing server, but
>>> this is
>>> > out of scope of current description).
>>> > Routing server routes calls to media servers based on logic that you
>>> can
>>> > easily define.
>>> > (Generally speaking, Routing server can route part of the calls onto
>>> itself,
>>> > but to simplify let's say that it will route only to Media Servers).
>>> >
>>> > Routing server is used to register subscribers (it will have directory
>>> files
>>> >
>>> ./freeswitch/conf/directory/default/10001whatever.xml-99999whatever.xml).
>>> > Media servers do NOT contain files with users (security issue? probably
>>> > should be managed on dialplan stage of media servers by comparing ip
>>> > addresses of requests).
>>> >
>>> > The whole routing can be presented by the following chain:
>>> > Call arrives into Default context-->RS default dialplan-->Lua
>>> script-->PHP
>>> > script to get destination MEDIA SERVER IP-->back to Lua script-->Bridge
>>> > "sofia/external/" to determined MS IP address-->[Bridge to fail over
>>> media
>>> > server, optionally]-->End
>>> > Let's say we need to route all international calls based on less loaded
>>> > server (other logic can be easily implemented in PHP script as well).
>>> > 1. RS Dialplan: default.xml
>>> > <include>
>>> >         <extension name="route">
>>> >                 <condition field="destination_number"
>>> > expression="^(00)(\d+)">
>>> >                         <action application="lua" data="route.lua $2"/>
>>> >                 </condition>
>>> >         </extension>
>>> > </include>
>>> >
>>> > 2. RS Lua script: route.lua
>>> > called_number = argv[1]; //without "00"
>>> > api = freeswitch.API();
>>> > server_ip_address = api:executeString("curl
>>> > http://localhost/get_server_ip.php);
>>> > forwarding_session_string =
>>> > "sofia/external/"..called_number.."@"..server_ip_address;
>>> > session:setVariable("bypass_media", "true"); //described below!!!
>>> > session:execute("bridge",forwarding_session_string);
>>> > 3. PHP script: get_server_ip.php
>>> > <<<  ANY PHP CODE THAT WILL QUERY COMMON DATABASE AND WILL DETERMINE
>>> LESS
>>> > LOADED AT THE MOMENT SERVER>>>
>>> > die($less_loaded_server_ip);
>>> >
>>> > All Media Servers should provide information about their current load
>>> into
>>> > one common database (dedicated server or routing server can be used for
>>> > database).
>>> > If you use scripts you can update this table on starting a call and
>>> after
>>> > call finished.
>>> > Or you can create a job that will query FreeSWITCH and will update the
>>> > database every, say, 10 seconds.
>>> >
>>> > Basically that's all (really!)
>>> > I beleive it can be used for most general purpose scenarios without
>>> need of
>>> > third-party tools.
>>> > P.S.
>>> > The only problem we had from the beginning was that RTP media traffic
>>> was
>>> > passing thru both servers: RS and Media Server.
>>> > But after we added the following command to the routing script, RTP
>>> media
>>> > traffic started to pass only thru Media server, making RS almost free.
>>> > session:setVariable("bypass_media", "true");
>>> >
>>> >
>>> > Best regards,
>>> > vIT
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From anthony.minessale at gmail.com  Tue Mar 16 06:45:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 08:45:29 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
Message-ID: <191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>

Are you bridging to sip or some ivr and making dingaling wait until the call
answers?

gtalk is sort of missing key telephony features like the concept of pdd and
indication progress etc.

Try answering the call in your dialplan before you bridge.

I think the gtalk client is getting impatient because you did not answer
soon enough with an ip choice
so it starts sending all these extra ones at you and gave up on the one we
chose.


On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > FS would prefer you run your kernel at 1000hz you are probably on a VM if
> > its 10000hz
> >
>
> It's a  physical machine. And I'm also running 20 skype clients. as
> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> 250 HZ. weird.
>
>
> here is the dingaling log with dl_debug on
>
> http://pastebin.freeswitch.org/12453
> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>
>
> still get this message on client:
>
> Sorry! The voice chat with language_lesson failed because of a network
> problem at 5:39 PM. Please try again.
>  Click here to report this to Google
>
>
>
> Thanks.
>
>
>
>
>
>
>
>
>
> >
> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> Windows client and web site on Mac. They both work on the other two
> >> instances of FreeSWITCH I mentioned in this thread.
> >>
> >> btw, can you give me a hint on the timer warning?
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > you would have to execute
> >> >
> >> > dl_debug on
> >> >
> >> > to get a trace that would tell us anything about that?
> >> >
> >> > Are you using the windows gtalk client or the web site?
> >> > We are following google who have been pushing for the online client
> >> > embedded
> >> > in the gmail website.
> >> >
> >> >
> >> >
> >> >
> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> here is the client:
> >> >>
> >> >> <profile type="client">
> >> >>  <param name="name" value="gmail.com"/>
> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >>  <param name="password" value="password"/>
> >> >>  <param name="server" value="talk.google.com"/>
> >> >>  <param name="dialplan" value="XML"/>
> >> >>  <param name="contex" value="public"/>
> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >>  <param name="rtp-ip" value="auto"/>
> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >>  <param name="auto-login" value="true"/>
> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >>  <param name="sasl" value="plain"/>
> >> >>  <param name="tls" value="true"/>
> >> >>  <param name="exten" value="5000"/>
> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> >> </profile>
> >> >>
> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> > ALL,
> >> >> >
> >> >> > I just upgrade FS to 16992.
> >> >> >
> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
> sounds
> >> >> > ok.
> >> >> >
> >> >> >
> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
> >> >> > detected!
> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> >> > experience audio problems.
> >> >> >
> >> >> > uname -a
> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05
> UTC
> >> >> > 2008 i686 GNU/Linux
> >> >> >
> >> >> >
> >> >> > I used a script to check the interupts
> >> >> >
> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >
> >> >> >
> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> answer.
> >> >> > I
> >> >> > originate a call from FS, the console blocks, gtalk client ring and
> >> >> > it
> >> >> > looks ok when I hit answer, but no sound, FS console still blocks.
> a
> >> >> > few seconds later gtalk client shows:
> >> >> >
> >> >> > Sorry! The voice chat with language_lesson failed because of a
> >> >> > network
> >> >> > problem at 5:50 PM. Please try again.
> >> >> >  Click here to report this to Google.
> >> >> >
> >> >> >
> >> >> > here is a log with bgapi:
> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >
> >> >> >
> >> >> > rev 14696 works on the same server same conf. another server
> running
> >> >> > 16958M works at the same time. I have no clue to check, is it
> related
> >> >> > to the timer as in 1)? can some help me to take a look? let me know
> >> >> > if
> >> >> > you need "dl_debug on" logs.
> >> >> >
> >> >> > Thanks.
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar 16 06:53:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 08:53:33 -0500
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
Message-ID: <5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>

Can you collect some logs and traces to show us this?

/b

On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:

> OK - sorry to make such a meal of this.  




From anthony.minessale at gmail.com  Tue Mar 16 06:54:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 08:54:33 -0500
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
Message-ID: <191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg
command itself
is designed to send a 1 way message to the channel, the reply is only to
confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only
thing you can do.

if you don't like async behaviour you could always send a blocking
instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000,
NULL);



On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  I agree, that is a possibility. However, when sending a ?SendMsg? with
> application ?break?, and receive ?+OK? as it?s reply, you kind of expect
> that it was received correctly, and will be handled in the event queue
> properly.
>
>
>
> I have a created a small patch that (probably) will fix this in FS, without
> breaking anything else. I just need to test it first ? if everything seems
> ok I will add a jira for it, with the patch attached.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 17:21
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could also wait for the execute_complete of the file you were playing
>
> On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
> On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> !DSPAM:4b9e60d032938314917241!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From peter.olsson at visionutveckling.se  Tue Mar 16 07:13:40 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 15:13:40 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>

Anthony, thanks for further information!

My goal is not to wait for the break to be executed, I just want to make sure it will be handled properly - and actually break the channel that is playing. So nothing wrong with the async behaviour itself - I like it the way it is.

I've been digging into the FS code for a couple of hours, and I think the problem occurs when I'm "fast" enough to add the break command, and then a new playback command in the event queue (and they both exist there). Then in the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked, switch_ivr_parse_all_events() will be called. If both events exist in the queue, the CF_BREAK will first be set, then on the next event it will be reset again, and the new playback will begin. But it will never break the original playback.

Since I havn't digging too deep in the code, I'm not 100% sure of this :) I was hoping you could help out with your expertice. My current patch looks something like this - do you think it's possible this causes other problems? By the way - sorry for getting this technical in the users-list, I guess this belongs more to the dev-list...

Index: src/switch_ivr.c
===================================================================
--- src/switch_ivr.c                         (revision 17000)
+++ src/switch_ivr.c                      (working copy)
@@ -660,12 +660,16 @@
 SWITCH_DECLARE(switch_status_t) switch_ivr_parse_all_events(switch_core_session_t *session)
 {
                             int x = 0;
+                           switch_channel_t *channel = switch_core_session_get_channel(session);

-
                             switch_ivr_parse_all_messages(session);

-                            while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS)
+                           while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS) {
                                                          x++;
+                                                        if (switch_channel_test_flag(channel, CF_BREAK)) {
+                                                                                     break;
+                                                        }
+        }

                             if (x) {
                                                          switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);

/Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 16 mars 2010 14:55
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg command itself
is designed to send a 1 way message to the channel, the reply is only to confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only thing you can do.

if you don't like async behaviour you could always send a blocking instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000, NULL);


On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
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FreeSWITCH Developer Conference
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
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ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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!DSPAM:4b9f8f4132931389420256!
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From moises.silva at gmail.com  Tue Mar 16 07:32:28 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 16 Mar 2010 10:32:28 -0400
Subject: [Freeswitch-users] SIP to PRI RECOVERY_ON_TIMER_EXPIRE error
	using Sangoma (sangoma_prid)
In-Reply-To: <10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>
References: <10128ef11003150645w2d7f46b1v5053960592490825@mail.gmail.com>
	<10128ef11003150651r793f61b9i91dadc169b97b907@mail.gmail.com>
	<4B9E4027.4000100@voiceworks.pl>
	<10128ef11003150721o546e5da8n79ff766c8722537@mail.gmail.com>
	<c4d05cbe1003150828m67f9fc56x67621913176e7ed4@mail.gmail.com>
	<10128ef11003152131o383e0b3el6b9628b74da56998@mail.gmail.com>
Message-ID: <c4d05cbe1003160732r509f534frafeea0921afb78f7@mail.gmail.com>

I'll contact you off-list to arrange ssh debug session.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Tue, Mar 16, 2010 at 12:31 AM, Tzury Bar Yochay <tzury.by at reguluslabs.com
> wrote:

>
>
> On Mon, Mar 15, 2010 at 5:28 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Ok, so the call is not even being placed in the ISDN network.
>>
>> Did you manage to get the sangoma_prid logs working? if that's the case
>> then paste them too, it seems sangoma_prid is not seeing the request from
>> FreeSWITCH to place a call and FreeSWITCH is timing out cuz there is no
>> response from sangoma_prid binary (they both communicate through sctp
>> socket).
>>
>
> This is the log (a result of `# /usr/sbin/sangoma_prid &>
> sangoma_prid.log`)
>
> ================System restart=============
> = Sangoma PRI Protocol Stack Daemon       =
> = Version: 1.58                           =
> = Date: Feb 18 2010                       =
> = Wanpipe Release: wanpipe-3.5.10.3          =
> = Revision:Revision: 15364          =
> ===========================================
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.overload.task
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.100%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.90%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.overload.task
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.100%-full
> DEBUG - AlarmRegistryImpl::addAlarm : Alarm is not unique :
> system.task.queue.90%-full
> WARN - detected an AFT-A101 board with an unsupported firmware version.
> (detected: 0x34, supported: 0x36). You may encounter unknown issues if you
> use an unsupported firmware version. Please contact support to get the
> proper instructions to get the supported firmware unless you have been
> instructed to use this firmware version.
> ERROR - Assertion failed (in_eNewState == STARTING_TSLINK_STATE)
> file=../../../stack/src/main.cpp line=134
> ERROR - ACE_Task::wait failed with error 22 (Invalid argument) . Task name
> = TelesoftStack
> ERROR - failed to waitUntilDone the TaskManager's thread TelesoftStack
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar 16 07:33:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 09:33:40 -0500
Subject: [Freeswitch-users] ESL and application "break" (to stop file
	playback) - timing issues.
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
Message-ID: <191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>

my last email had a code suggestion

On Tue, Mar 16, 2010 at 9:13 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Anthony, thanks for further information!
>
>
>
> My goal is not to wait for the break to be executed, I just want to make
> sure it will be handled properly ? and actually break the channel that is
> playing. So nothing wrong with the async behaviour itself - I like it the
> way it is.
>
>
>
> I?ve been digging into the FS code for a couple of hours, and I think the
> problem occurs when I?m ?fast? enough to add the break command, and then a
> new playback command in the event queue (and they both exist there). Then in
> the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked,
> switch_ivr_parse_all_events() will be called. If both events exist in the
> queue, the CF_BREAK will first be set, then on the next event it will be
> reset again, and the new playback will begin. But it will never break the
> original playback.
>
>
>
> Since I havn?t digging too deep in the code, I?m not 100% sure of this :) I
> was hoping you could help out with your expertice. My current patch looks
> something like this ? do you think it?s possible this causes other problems?
> By the way ? sorry for getting this technical in the users-list, I guess
> this belongs more to the dev-list...
>
>
>
> Index: src/switch_ivr.c
>
> ===================================================================
>
> --- src/switch_ivr.c                         (revision 17000)
>
> +++ src/switch_ivr.c                      (working copy)
>
> @@ -660,12 +660,16 @@
>
>  SWITCH_DECLARE(switch_status_t)
> switch_ivr_parse_all_events(switch_core_session_t *session)
>
>  {
>
>                              int x = 0;
>
> +                           switch_channel_t *channel =
> switch_core_session_get_channel(session);
>
>
>
> -
>
>                              switch_ivr_parse_all_messages(session);
>
>
>
> -                            while (switch_ivr_parse_next_event(session) ==
> SWITCH_STATUS_SUCCESS)
>
> +                           while (switch_ivr_parse_next_event(session) ==
> SWITCH_STATUS_SUCCESS) {
>
>                                                           x++;
>
> +                                                        if
> (switch_channel_test_flag(channel, CF_BREAK)) {
>
> +
> break;
>
> +                                                        }
>
> +        }
>
>
>
>                              if (x) {
>
>
> switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);
>
>
>
> /Peter Olsson
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 16 mars 2010 14:55
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> These events sent via sendmsg are designed to be asynchronous.
> artificial waiting may be adding to the problem.
>
> Step outside of what you are trying to do and consider that the sendmsg
> command itself
> is designed to send a 1 way message to the channel, the reply is only to
> confirm that it was sent not the reply from
> the application you are executing, because executing an app is not the only
> thing you can do.
>
> if you don't like async behaviour you could always send a blocking
> instruction like the read or play_and_get_digits app to collect your info.
>
>
> If anything, it might be ok to put this in the "break" app at the end
>
> switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000,
> NULL);
>
>
>  On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I agree, that is a possibility. However, when sending a ?SendMsg? with
> application ?break?, and receive ?+OK? as it?s reply, you kind of expect
> that it was received correctly, and will be handled in the event queue
> properly.
>
>
>
> I have a created a small patch that (probably) will fix this in FS, without
> breaking anything else. I just need to test it first ? if everything seems
> ok I will add a jira for it, with the patch attached.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 17:21
>
>
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could also wait for the execute_complete of the file you were playing
>
> On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> Thanks for the reply.
>
>
>
> Yes, I thought of this.
>
>
>
> But that still shouldn?t solve the issue? It will only tell that the
> CF_BREAK flag was set on the channel, not that the current playback detected
> the flag in the loop (which I guess is done in a different thread), or is
> this event sent once the playback really did stop?
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 15 mars 2010 16:31
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] ESL and application "break" (to stop file
> playback) - timing issues.
>
>
>
> you could wait for the execute_complete event for your break command.
>
> On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I?m not sure if this belongs in Jira, or if it?s possible to do it another
> way ? so I try the list first :)
>
>
>
> I?m using ESL to do some IVR functions. Basically I play some sound files,
> wait for DTMF, and then do something, like record a message, transfer the
> call to a mobile phone etc. Quite basic and simple.
>
>
>
> I?ve noticed one problem with this, and it?s when I want to stop playback
> using command ?break?. In 99% of the time everything works as expected, but
> sometimes I get this problem ? and the problem is that the current file
> doesn?t stop playing. And I think I know why this is happening.
>
>
>
> Lets say I first send playback(tone_stream://%(1500, 3500, 440.0,
> 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and
> loop through it 100 times.. When I want to stop this tone I send the break
> command, and directly after this a new playback command. Maybe 1 time in 100
> tries the tone_stream doesn?t stop playing. I think this is becuase how
> break works. From what I understand from the source it just sets the
> CF_BREAK flag on the channel, and then the playback application will detect
> this and stop the playback. However, when I send a new playback command
> immediately after break it will reset the CF_BREAK-flag again, to make sure
> it won?t cause any problems for the new playback. So if the original
> playback (tone_stream) didn?t check the flag before it was reset, it won?t
> known that it was supposed to stop playing.
>
>
>
> I guess I could wait 50ms before sending the new playback message, but I
> can really never be sure that it was handled properly.
>
>
>
> Is there any other way around this? How about ?break <uuid> all?, will it
> make any difference in this case? Or should this just be treated as a timing
> bug, and filed to Jira? Any help on this would be greatly appreciated.
>
>
>
> These are the messages I?m sending;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100
>
>
>
> When I don?t want this to play anymore I send a break;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: break
>
>
>
> And immediately after this I want to play another file;
>
>
>
> SendMsg <uuid>
>
> call-command: execute
>
> execute-app-name: playback
>
> execute-app-arg: file/to/play.wav
>
>
>
>
>
> Regards,
>
>
>
> Peter Olsson
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
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>
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From peter.olsson at visionutveckling.se  Tue Mar 16 08:00:14 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 16 Mar 2010 16:00:14 +0100
Subject: [Freeswitch-users] ESL and application "break" (to stop
	file	playback) - timing issues.
In-Reply-To: <191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5577434BCE@cooper>
	<191c3a031003150831p42041ceeu5e1b12bbee1d9b1c@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577434DE9@cooper>
	<191c3a031003150921wf2560cdj74a91b36dc70eb71@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5577435072@cooper>
	<191c3a031003160654g2126c97cnd658d80638849437@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557743513F@cooper>
	<191c3a031003160733j5c5640f4u7bb51c7a2f744966@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5577435208@cooper>

Yes, I noticed that - thanks again :)

What exactly would that line do, I dont know how the CF_BROADCAST flag works.. What exacly would it wait for?

Just one more question about my patch - what I'm trying to do is to make sure that CF_BREAK flag is handled in switch_ivr_play_file() (and others calling switch_ivr_parse_all_events() in the loop), even when the event queue holds more than one event (first event "break", and next one "playback", which will reset CF_BREAK flag again). As it works right now CF_BREAK will not be handled in the first playback loop, if the queue holds a break event and then a new playback - is that really the way it should be?

Sorry for all questions, I just want know more, and understand the FS code a bite more...

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 16 mars 2010 15:34
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

my last email had a code suggestion
On Tue, Mar 16, 2010 at 9:13 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Anthony, thanks for further information!

My goal is not to wait for the break to be executed, I just want to make sure it will be handled properly - and actually break the channel that is playing. So nothing wrong with the async behaviour itself - I like it the way it is.

I've been digging into the FS code for a couple of hours, and I think the problem occurs when I'm "fast" enough to add the break command, and then a new playback command in the event queue (and they both exist there). Then in the main loop of switch_ivr_play_file(), after CF_BREAK flag is checked, switch_ivr_parse_all_events() will be called. If both events exist in the queue, the CF_BREAK will first be set, then on the next event it will be reset again, and the new playback will begin. But it will never break the original playback.

Since I havn't digging too deep in the code, I'm not 100% sure of this :) I was hoping you could help out with your expertice. My current patch looks something like this - do you think it's possible this causes other problems? By the way - sorry for getting this technical in the users-list, I guess this belongs more to the dev-list...

Index: src/switch_ivr.c
===================================================================
--- src/switch_ivr.c                         (revision 17000)
+++ src/switch_ivr.c                      (working copy)
@@ -660,12 +660,16 @@
 SWITCH_DECLARE(switch_status_t) switch_ivr_parse_all_events(switch_core_session_t *session)
 {
                             int x = 0;
+                           switch_channel_t *channel = switch_core_session_get_channel(session);

-
                             switch_ivr_parse_all_messages(session);

-                            while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS)
+                           while (switch_ivr_parse_next_event(session) == SWITCH_STATUS_SUCCESS) {
                                                          x++;
+                                                        if (switch_channel_test_flag(channel, CF_BREAK)) {
+                                                                                     break;
+                                                        }
+        }

                             if (x) {
                                                          switch_ivr_sleep(session, 0, SWITCH_TRUE, NULL);

/Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 16 mars 2010 14:55

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

These events sent via sendmsg are designed to be asynchronous.
artificial waiting may be adding to the problem.

Step outside of what you are trying to do and consider that the sendmsg command itself
is designed to send a 1 way message to the channel, the reply is only to confirm that it was sent not the reply from
the application you are executing, because executing an app is not the only thing you can do.

if you don't like async behaviour you could always send a blocking instruction like the read or play_and_get_digits app to collect your info.


If anything, it might be ok to put this in the "break" app at the end

switch_channel_wait_for_flag(channel, CF_BROADCAST, SWITCH_FALSE, 2000, NULL);

On Tue, Mar 16, 2010 at 6:41 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I agree, that is a possibility. However, when sending a "SendMsg" with application "break", and receive "+OK" as it's reply, you kind of expect that it was received correctly, and will be handled in the event queue properly.

I have a created a small patch that (probably) will fix this in FS, without breaking anything else. I just need to test it first - if everything seems ok I will add a jira for it, with the patch attached.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 17:21

Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could also wait for the execute_complete of the file you were playing
On Mon, Mar 15, 2010 at 9:40 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Thanks for the reply.

Yes, I thought of this.

But that still shouldn't solve the issue? It will only tell that the CF_BREAK flag was set on the channel, not that the current playback detected the flag in the loop (which I guess is done in a different thread), or is this event sent once the playback really did stop?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Anthony Minessale
Skickat: den 15 mars 2010 16:31
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] ESL and application "break" (to stop file playback) - timing issues.

you could wait for the execute_complete event for your break command.
On Mon, Mar 15, 2010 at 3:20 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I'm not sure if this belongs in Jira, or if it's possible to do it another way - so I try the list first :)

I'm using ESL to do some IVR functions. Basically I play some sound files, wait for DTMF, and then do something, like record a message, transfer the call to a mobile phone etc. Quite basic and simple.

I've noticed one problem with this, and it's when I want to stop playback using command "break". In 99% of the time everything works as expected, but sometimes I get this problem - and the problem is that the current file doesn't stop playing. And I think I know why this is happening.

Lets say I first send playback(tone_stream://%(1500, 3500, 440.0, 0.0);loops=100) to the channel. This will play a ring-tone (swedish) and loop through it 100 times.. When I want to stop this tone I send the break command, and directly after this a new playback command. Maybe 1 time in 100 tries the tone_stream doesn't stop playing. I think this is becuase how break works. From what I understand from the source it just sets the CF_BREAK flag on the channel, and then the playback application will detect this and stop the playback. However, when I send a new playback command immediately after break it will reset the CF_BREAK-flag again, to make sure it won't cause any problems for the new playback. So if the original playback (tone_stream) didn't check the flag before it was reset, it won't known that it was supposed to stop playing.

I guess I could wait 50ms before sending the new playback message, but I can really never be sure that it was handled properly.

Is there any other way around this? How about "break <uuid> all", will it make any difference in this case? Or should this just be treated as a timing bug, and filed to Jira? Any help on this would be greatly appreciated.

These are the messages I'm sending;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(1500, 3500, 440.0, 0.0);loops=100

When I don't want this to play anymore I send a break;

SendMsg <uuid>
call-command: execute
execute-app-name: break

And immediately after this I want to play another file;

SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: file/to/play.wav


Regards,

Peter Olsson


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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
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From jbrucehopkins at gmail.com  Tue Mar 16 08:05:54 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 15:05:54 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
	<5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
	<d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>
Message-ID: <d03a10e01003160805t36c4f8dbhedaecaf0626df9ae@mail.gmail.com>

... and here are two more Wireshark traces showing the successfull call
scenario, where phone A is configured to use only Sppex (rather than using
Alaw and requiring Freeswitch to transcode).

thanks again
Bruce

On 16 March 2010 14:41, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Sure.  I put the Freeswitch log in pastebin.  here are two Wireshark
> captures in the condition where the call fails- run on two PC's running
> Eyebeam clients.  Phone A initiates the call with only PCMA enabled.  Phone
> B has only Speex enabled.
>
> Thanks for taking the time to look at these.
> Bruce
>
>
> On 16 March 2010 13:53, Brian West <brian at freeswitch.org> wrote:
>
>> Can you collect some logs and traces to show us this?
>>
>> /b
>>
>> On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:
>>
>> > OK - sorry to make such a meal of this.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From ivdreg at gmail.com  Tue Mar 16 08:20:58 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 16 Mar 2010 17:20:58 +0200
Subject: [Freeswitch-users] Mod_limit questions
Message-ID: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>

Hi All,

I have some questions regarding mod_limit.

1. I'm using 16674 in production but tested and with 17001. I try mod_limit
with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash
never tries to write data to SQL. It works (counts sessions) on log screen
but no data in MySQL/Sqlite. It seems that data is cached but cache never
flushed to DB. Is anyone has this problem. My limits are:

<action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
${destination_number} 5 !INCOMING_CALL_BARRED" />
<action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />

and they are applied on inbound call profile in same extension.


2. I need to implement rule maximum of 50 calls per day/24h from the same
CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
example "Limit calls to 5 calls every 10 minutes" is about concurrent call
but not calls processed as count. Any idea ?

Thanks for any help
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From tculjaga at gmail.com  Tue Mar 16 08:38:05 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 16:38:05 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
Message-ID: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Tihomir,
>            Could you please add more details to the following?
>
>            "So, i'd suggest to bring up a LAN_HA and have a floating
> address between your routing servers.... of course the you should be using
> ODBC in the core to have registrations on both servers."
>
> Do you have any links that describe how to implement this kind of fail
> over?
>
>
This is the topic for you:
http://www.ultramonkey.org/3/topologies/
and this is what i already have implemented:
http://www.ultramonkey.org/3/topologies/ha-eg.html




> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
> will go down, traffic will be addressed to another, but at the moment I
> don't have enough knowledge how to implement this.
>
>
don't like the idea as you are introducing one more point of failure :)


> Thanks,
> vIT
>
>
>
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From tculjaga at gmail.com  Tue Mar 16 08:38:47 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 16:38:47 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
Message-ID: <65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:19 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi Tihomir,
> What do you mean by "took from Mera"?
> If someone else already implemented this, could you please provide me a
> link?
>
>
>
>
read the file attached :)


T.
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From anthony.minessale at gmail.com  Tue Mar 16 08:48:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 10:48:20 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
Message-ID: <191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>

depending on the phone FreeSWITCH will update the display any time a call is
bridged.

it works on polycom,snom and aastra and I am not sure if there are more.


On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:

> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
>
> When a call is received, the extension really needs to display information
> about the origin and the intended recipient of the call. Normally for an
> incoming external call, the former is supplied by the 'incoming CLI' (often
> linked to locally stored name) and the latter probably by associating a name
> with each incoming line, so in this way the recipient can see at a glance
> who called whom (or what).
>
> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
>
> So, how does FS handle all the above?
> What does it pass to the recipient, for both external and internal calls?
> Is this part of FS configuration? IOW, do you have programmatic access to
> such information and can you specify exactly what is sent to an extension?
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From srinivas.ksvreddy at gmail.com  Tue Mar 16 08:49:03 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Tue, 16 Mar 2010 21:19:03 +0530
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
Message-ID: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>

we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to
use new versions.

srinivas

On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> Hi,
>
> i am using freeswitch1.0.2, farwarding call from freeswitch to cisco
> like just farwading the call.
> <action application="bridge" data=
> sofia/internal/${destination_number}@ciscogatewayip<sofia/internal/$%7Bdestination_number%7D at ciscogatewayip>
> / <data=sofia/internal/$%7Bdestination_number%7D at ciscogatewayip/>>
> i do not using any gateway here, just farwarding call to router, is this
> correct way of doing? any idea
>
> --
> Srinivasula Reddy K
>



-- 
Srinivasula Reddy K
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From vetali100 at gmail.com  Tue Mar 16 08:51:19 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 17:51:19 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
Message-ID: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>

Using it you still have one point of failure - only one router and it's on
one internet line...

What if one of these things goes down for a period? :-)

I would prefer to have 2 separate servers on 2 different internet lines
(better different providers even).

Thanks,
vIT


2010/3/16 Tihomir Culjaga <tculjaga at gmail.com>

>
>
> On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Tihomir,
>>            Could you please add more details to the following?
>>
>>            "So, i'd suggest to bring up a LAN_HA and have a floating
>> address between your routing servers.... of course the you should be using
>> ODBC in the core to have registrations on both servers."
>>
>> Do you have any links that describe how to implement this kind of fail
>> over?
>>
>>
> This is the topic for you:
> http://www.ultramonkey.org/3/topologies/
> and this is what i already have implemented:
> http://www.ultramonkey.org/3/topologies/ha-eg.html
>
>
>
>
>> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
>> will go down, traffic will be addressed to another, but at the moment I
>> don't have enough knowledge how to implement this.
>>
>>
> don't like the idea as you are introducing one more point of failure :)
>
>
>> Thanks,
>> vIT
>>
>>
>>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Tue Mar 16 08:51:19 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 17:51:19 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
Message-ID: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>

Using it you still have one point of failure - only one router and it's on
one internet line...

What if one of these things goes down for a period? :-)

I would prefer to have 2 separate servers on 2 different internet lines
(better different providers even).

Thanks,
vIT


2010/3/16 Tihomir Culjaga <tculjaga at gmail.com>

>
>
> On Tue, Mar 16, 2010 at 2:31 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Tihomir,
>>            Could you please add more details to the following?
>>
>>            "So, i'd suggest to bring up a LAN_HA and have a floating
>> address between your routing servers.... of course the you should be using
>> ODBC in the core to have registrations on both servers."
>>
>> Do you have any links that describe how to implement this kind of fail
>> over?
>>
>>
> This is the topic for you:
> http://www.ultramonkey.org/3/topologies/
> and this is what i already have implemented:
> http://www.ultramonkey.org/3/topologies/ha-eg.html
>
>
>
>
>> I thought about configuring 2 DNS servers on 2 Routing servers, so if one
>> will go down, traffic will be addressed to another, but at the moment I
>> don't have enough knowledge how to implement this.
>>
>>
> don't like the idea as you are introducing one more point of failure :)
>
>
>> Thanks,
>> vIT
>>
>>
>>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 16 09:00:05 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:00:05 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <DC993C45F9704BE585B4D3A344911A31@MOBILEE1705>

so share your customized version to the community and maybe
we ll be glad to help you.... this is the principle of open source
  ----- Original Message ----- 
  From: srinivasula reddy 
  To: freeswitch-users 
  Sent: Tuesday, March 16, 2010 11:49 AM
  Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


  we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.

  srinivas


  On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:

    Hi,

    i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. 
    <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>
    i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea

    -- 
    Srinivasula Reddy K




  -- 
  Srinivasula Reddy K



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From msc at freeswitch.org  Tue Mar 16 09:01:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 09:01:36 -0700
Subject: [Freeswitch-users] Set/Export channel variable [SOLVED]
In-Reply-To: <4295808702047090347@unknownmsgid>
References: <D3A07054-9C26-4504-AE2E-C4F9882BA421@freeswitch.org>
	<8259813661645760166@unknownmsgid>
	<87f2f3b91003151128g656519bcq3d79516e5309effc@mail.gmail.com>
	<4295808702047090347@unknownmsgid>
Message-ID: <87f2f3b91003160901x282ab10x6891be46462401d2@mail.gmail.com>

On Mon, Mar 15, 2010 at 11:21 PM, Ahmed Ajmal <ahmed.ajmal at breezecom.ae>wrote:

>  So it seems that the FS console by default only displays the ?a? leg vars
> which is why I couldn?t see it (and it makes sense too since gateway is on
> the b leg side). I have the ESL script now display both the inbound/outbound
> after which I can see the sip_gateway_name var.
>
Good catch!
-MC
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From msc at freeswitch.org  Tue Mar 16 08:59:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 08:59:58 -0700
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>

On Tue, Mar 16, 2010 at 8:49 AM, srinivasula reddy <
srinivas.ksvreddy at gmail.com> wrote:

> we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to
> use new versions.
>
>
Bummer. Unfortunately you are in a precarious situation because no one here
has used 1.0.2 for more than a year and the devs are not exactly fond of
that particular revision. The good news is that FS is very modular, so
unless you've been messing around with the core or mod_sofia you might be
able to update to 1.0.5 and reapply your customization.
-MC
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From brian at freeswitch.org  Tue Mar 16 08:59:53 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 10:59:53 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
Message-ID: <C4A9FC2C-A475-4557-BA52-13A734BC75A3@freeswitch.org>

At any time did you happen to think ... "I should join the project and help out."?

/b

On Mar 16, 2010, at 10:49 AM, srinivasula reddy wrote:

> we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.
>  
> srinivas




From msc at freeswitch.org  Tue Mar 16 09:04:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 09:04:14 -0700
Subject: [Freeswitch-users] answering call in irly_media mode
In-Reply-To: <4B9F3A97.4030609@gmail.com>
References: <4B9F3A97.4030609@gmail.com>
Message-ID: <87f2f3b91003160904q59aa4cd3u884b85e3d3ec312f@mail.gmail.com>

On Tue, Mar 16, 2010 at 1:00 AM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

> hi all
> i want to setup a custom ringback tone for inbound (not outbound calls)
> is in irly_media mode but i still heare the normal US ringback.
> where is my dialplan extension:
> http://pastebin.freeswitch.org/12452
> thanks
>
> Check out the ringback variable:
http://wiki.freeswitch.org/wiki/Channel_Variables#ringback

-MC
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From rupa at rupa.com  Tue Mar 16 08:58:21 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 10:58:21 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
Message-ID: <b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>

1) limit_hash use an in-memory hash for it's storage. There is nor storing
to a database when using limit_hash.  If you want to use the database, then
use plain limit.

2) If you want a max calls in a 24hrs period use you rate with limit_hash.
 This should be about a true max per period, not concurrent.

limit_hash realm id 50/86400

should do what you want -- from memory.  double check the wiki/help to
verify syntax.

On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Hi All,
>
> I have some questions regarding mod_limit.
>
> 1. I'm using 16674 in production but tested and with 17001. I try mod_limit
> with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash
> never tries to write data to SQL. It works (counts sessions) on log screen
> but no data in MySQL/Sqlite. It seems that data is cached but cache never
> flushed to DB. Is anyone has this problem. My limits are:
>
> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
> ${destination_number} 5 !INCOMING_CALL_BARRED" />
> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>
> and they are applied on inbound call profile in same extension.
>
>
> 2. I need to implement rule maximum of 50 calls per day/24h from the same
> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
> but not calls processed as count. Any idea ?
>
> Thanks for any help
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Tue Mar 16 09:09:25 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 11:09:25 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>
References: <f8af5741003150836p5db33ae8t7b33b808def4d6a8@mail.gmail.com>
	<f8af5741003160849pa980de5y7ab4b956de23674d@mail.gmail.com>
	<87f2f3b91003160859q4cb7d79asac327aa5b0019423@mail.gmail.com>
Message-ID: <B6737CFC-B597-4433-B99B-F8F12DD6F3F5@freeswitch.org>

1.0.2 was used for exactly two weeks before 1.0.3 came out.. 1.0.2 has a boat load of bugs.

/b

On Mar 16, 2010, at 10:59 AM, Michael Collins wrote:

> Bummer. Unfortunately you are in a precarious situation because no one here has used 1.0.2 for more than a year and the devs are not exactly fond of that particular revision. The good news is that FS is very modular, so unless you've been messing around with the core or mod_sofia you might be able to update to 1.0.5 and reapply your customization. 
> -MC




From ken at ukgb.net  Tue Mar 16 09:23:51 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 16:23:51 +0000
Subject: [Freeswitch-users] call info display
In-Reply-To: <191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
Message-ID: <3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>

Thanks, but could I ask for more specifics? Exactly what info is sent and is it sent as part of the SIP info, i.e. so any SIP device ought to display it correctly? Are there any docs I can read that will explain in detail about the information that is sent (i.e. call info, not SIP protocol stuff) and how to configureFS to send exactly what is required?


On 16 Mar 2010, at 15:48, Anthony Minessale wrote:

> depending on the phone FreeSWITCH will update the display any time a call is bridged.
> 
> it works on polycom,snom and aastra and I am not sure if there are more.
> 
> 
> On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
>> Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.
>> 
>> When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).
>> 
>> Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?
>> 
>> So, how does FS handle all the above?
>> What does it pass to the recipient, for both external and internal calls?
>> Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?
> 


Ken G i l l e t t

_/_/_/_/_/_/_/_/






From vetali100 at gmail.com  Tue Mar 16 09:28:52 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Tue, 16 Mar 2010 18:28:52 +0200
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
	<65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
Message-ID: <a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>

I have read the document and except that it describes one signalling server
and three media servers with routing to less busiest server it has nothing
else related to solution I wrote for FreeSWITCH.
Basically topology and high level part will be repeated in any solutions,
because this is obvious for everyone.
The same topology as described in this document and by me will be created
using any routing server (OpenSIPS or whatever) and will come to the mind of
most people that need to implement load balancing.


Regards,
vIT



2010/3/16 Tihomir Culjaga <tculjaga at gmail.com>

>
>
> On Tue, Mar 16, 2010 at 2:19 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi Tihomir,
>> What do you mean by "took from Mera"?
>> If someone else already implemented this, could you please provide me a
>> link?
>>
>>
>>
>>
> read the file attached :)
>
>
> T.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 16 09:34:36 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:34:36 -0400
Subject: [Freeswitch-users] modules update
Message-ID: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>

Hi there,

do I need to "make install" separatly modules (like distributor)
everytime I upgrade from trunk (make current) ?

Thanks

Franck
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From testeador01 at gmail.com  Tue Mar 16 09:39:34 2010
From: testeador01 at gmail.com (Milena)
Date: Tue, 16 Mar 2010 11:39:34 -0500
Subject: [Freeswitch-users] modules update
Message-ID: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>

Hello,

if you edit modules.conf (in the source folder) and uncomment every module
that you want to build, you will not need to make install every module, make
current is enough.

:)
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From brian at freeswitch.org  Tue Mar 16 09:41:28 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 11:41:28 -0500
Subject: [Freeswitch-users] modules update
In-Reply-To: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
References: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
Message-ID: <1BBBD2D5-577E-401D-9560-96DA87CDA34A@freeswitch.org>

Make sure modules.conf has the module in it and type "make current" it will rebuild everything and update you to the latest SVN.

/b

On Mar 16, 2010, at 11:34 AM, Madovsky wrote:

> Hi there,
>  
> do I need to "make install" separatly modules (like distributor)
> everytime I upgrade from trunk (make current) ?
>  
> Thanks
>  
> Franck

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From rupa at rupa.com  Tue Mar 16 09:43:14 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 11:43:14 -0500
Subject: [Freeswitch-users] modules update
In-Reply-To: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
References: <499711A1E12745C2BE847A42033CB725@MOBILEE1705>
Message-ID: <b39273531003160943r3b89c3fbg91c915649b91ee02@mail.gmail.com>

make current

is enough if your modules.conf has the modules you are interested
uncommented.

On Tue, Mar 16, 2010 at 11:34 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi there,
>
> do I need to "make install" separatly modules (like distributor)
> everytime I upgrade from trunk (make current) ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From infos at madovsky.org  Tue Mar 16 09:48:37 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 12:48:37 -0400
Subject: [Freeswitch-users] modules update
References: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>
Message-ID: <99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>

Ok I was not unsure since on wiki (for distributor module)
it says "make mod_distributor" and "make mod_distributor-install"
to install it properly.

thanks !
  ----- Original Message ----- 
  From: Milena 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:39 PM
  Subject: Re: [Freeswitch-users] modules update


  Hello,


  if you edit modules.conf (in the source folder) and uncomment every module that you want to build, you will not need to make install every module, make current is enough.


  :)


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From tculjaga at gmail.com  Tue Mar 16 09:58:51 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 17:58:51 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<5fe6fa8f1003160613o4b54df83g5da2ce7b3fd4b3e7@mail.gmail.com>
	<a858e7691003160621s3d2b60fcp26533d1dbb20a77c@mail.gmail.com>
	<a858e7691003160631m4c08737el72d4e6b617cce06e@mail.gmail.com>
	<65d96fc81003160838p272fca54hcdec8c9fb9860856@mail.gmail.com>
	<a858e7691003160851w53594fa5l25168cb42e07f759@mail.gmail.com>
Message-ID: <65d96fc81003160958n274d7a18r4b952dcf451d9202@mail.gmail.com>

On Tue, Mar 16, 2010 at 4:51 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Using it you still have one point of failure - only one router and it's on
> one internet line...
>
> What if one of these things goes down for a period? :-)
>
>
well, thats something out of our reach ... we are taking care about the
application and services that are working on top.... well the network is a
totally different topic and it has to be addressed with care.


> I would prefer to have 2 separate servers on 2 different internet lines
> (better different providers even).
>
>
right ... with BGP/MPLS in between as well ... as i said it is a different
topic :)


> Thanks,
> vIT
>
>
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From tculjaga at gmail.com  Tue Mar 16 10:00:56 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Mar 2010 18:00:56 +0100
Subject: [Freeswitch-users] Load balancing solution using pure FreeSWITCH
In-Reply-To: <a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>
References: <a858e7691003160504w76703481x1b48e1121cf0567@mail.gmail.com>
	<65d96fc81003160607g5d77c8b3s52fb5a018aa82bdb@mail.gmail.com>
	<a858e7691003160619n11d131bbm2b7a7427e3c5d015@mail.gmail.com>
	<65d96fc81003160838p2cc2575et86de60faedbc3415@mail.gmail.com>
	<a858e7691003160928x59181e42u3184801dc514f772@mail.gmail.com>
Message-ID: <65d96fc81003161000g29703556je24c88cace064e2a@mail.gmail.com>

On Tue, Mar 16, 2010 at 5:28 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> I have read the document and except that it describes one signalling server
> and three media servers with routing to less busiest server it has nothing
> else related to solution I wrote for FreeSWITCH.
> Basically topology and high level part will be repeated in any solutions,
> because this is obvious for everyone.
> The same topology as described in this document and by me will be created
> using any routing server (OpenSIPS or whatever) and will come to the mind of
> most people that need to implement load balancing.
>
>
> Regards,
> vIT
>
>
> it is just the topology that is the same ... RS + MS, after that you are on
your own... that's what i meant.

T.
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From anthony.minessale at gmail.com  Tue Mar 16 10:11:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 12:11:56 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
Message-ID: <191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>

as i said it is phone specific.

The callee id and caleeid number is sent

on polycom and astra via sip UPDATE
on snom and FS->FS via INFO


There is no configuration necessary, its the default behavior.


On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:

> Thanks, but could I ask for more specifics? Exactly what info is sent and
> is it sent as part of the SIP info, i.e. so any SIP device ought to display
> it correctly? Are there any docs I can read that will explain in detail
> about the information that is sent (i.e. call info, not SIP protocol stuff)
> and how to configureFS to send exactly what is required?
>
>
> On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
>
> > depending on the phone FreeSWITCH will update the display any time a call
> is bridged.
> >
> > it works on polycom,snom and aastra and I am not sure if there are more.
> >
> >
> > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> >> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
> >>
> >> When a call is received, the extension really needs to display
> information about the origin and the intended recipient of the call.
> Normally for an incoming external call, the former is supplied by the
> 'incoming CLI' (often linked to locally stored name) and the latter probably
> by associating a name with each incoming line, so in this way the recipient
> can see at a glance who called whom (or what).
> >>
> >> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
> >>
> >> So, how does FS handle all the above?
> >> What does it pass to the recipient, for both external and internal
> calls?
> >> Is this part of FS configuration? IOW, do you have programmatic access
> to such information and can you specify exactly what is sent to an
> extension?
> >
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From william.suffill at gmail.com  Tue Mar 16 11:29:32 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 14:29:32 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003141811.58645.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>

FSComm interfaces with sound devices via portaudio at this point. If
portaudio can't use your devices correctly that would cause it not to work
in any portaudio application including FSComm.

What version of Fedora? I'm sure this is fixable but really depends more on
the underlying system more than code in FSComm.


-- W
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From msc at freeswitch.org  Tue Mar 16 11:32:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 11:32:17 -0700
Subject: [Freeswitch-users] modules update
In-Reply-To: <99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>
References: <c29b930b1003160939q26cf779ar5fd512dd53103f32@mail.gmail.com>
	<99357C2C0F0A4E1AA93718008B708241@MOBILEE1705>
Message-ID: <87f2f3b91003161132y7f377c82jed099fd4d93aa0d8@mail.gmail.com>

On Tue, Mar 16, 2010 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:

>  Ok I was not unsure since on wiki (for distributor module)
> it says "make mod_distributor" and "make mod_distributor-install"
> to install it properly.
>
> thanks !
>

You can do "make mod_<module-name>-install" for any module. It's handy if
you, for example, just did a make current and realized that you hadn't built
mod_distributor. You can edit modules.conf but you don't have to run through
a whole make current again...

-MC

>  ----- Original Message -----
> *From:* Milena <testeador01 at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 12:39 PM
> *Subject:* Re: [Freeswitch-users] modules update
>
> Hello,
>
> if you edit modules.conf (in the source folder) and uncomment every module
> that you want to build, you will not need to make install every module, make
> current is enough.
>
> :)
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ken at ukgb.net  Tue Mar 16 11:52:34 2010
From: ken at ukgb.net (Ken Gillett)
Date: Tue, 16 Mar 2010 18:52:34 +0000
Subject: [Freeswitch-users] call info display
In-Reply-To: <191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
Message-ID: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>

Previously I had a company with a call centre, in which we used a Meridian Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of DDI numbers. Each client was provided with one of those DDI numbers on which their customers would call us, so it was vital the operator knew which client's customer was calling. Within the configuration of the Option 11 I named each incoming DDI with the respective client's (Company) name and then that name was sent to the extension, along with the incoming CLI, so each operator knew which client's number had been called (and hence how to respond) and also who was calling, either their number, or name if it could be matched in an address directory (although cannot now remember if that was done by the switch itself or the extension phone).

Nothing strange about any of that, but I'm trying to relate those same requirements to the capabilities of FS. My assumption has been that FS would be able to match anything traditional telephone switches could manage, but I now need some confirmation of this.

So, are the callee ID and number the incoming line name and number respectively and it's those that FS sends to the extension for display?

Where does this data come from? Surely naming the incoming lines must be part of the configuration.

Can I name them to whatever I want?

Is the incoming CLI number sent directly to the extension (where it's up to the extension what it then displays)?
Or
Does FS do any matching to a name in a directory first and then send the matched name to the extension?

Sorry to keep asking - I'm sure this is very obvious to those who know, but I'm still trying to learn all about FS and what it can and can't do.


On 16 Mar 2010, at 17:11, Anthony Minessale wrote:

> as i said it is phone specific.
> 
> The callee id and caleeid number is sent 
> 
> on polycom and astra via sip UPDATE
> on snom and FS->FS via INFO
> 
> 
> There is no configuration necessary, its the default behavior.
> 
> 
> On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:
> Thanks, but could I ask for more specifics? Exactly what info is sent and is it sent as part of the SIP info, i.e. so any SIP device ought to display it correctly? Are there any docs I can read that will explain in detail about the information that is sent (i.e. call info, not SIP protocol stuff) and how to configureFS to send exactly what is required?
> 
> 
> On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
> 
> > depending on the phone FreeSWITCH will update the display any time a call is bridged.
> >
> > it works on polycom,snom and aastra and I am not sure if there are more.
> >
> >
> > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> >> Sorry to ask what are perhaps very basic questions, but could someone please enlighten me on FreeSwitch's capabilities with regard to information for display on the recipient's extension.
> >>
> >> When a call is received, the extension really needs to display information about the origin and the intended recipient of the call. Normally for an incoming external call, the former is supplied by the 'incoming CLI' (often linked to locally stored name) and the latter probably by associating a name with each incoming line, so in this way the recipient can see at a glance who called whom (or what).
> >>
> >> Am I right in thinking that within SIP there is the provision to send information about the calling party (origin), so the called party (recipient) doesn't have to rely on CLI and matching numbers to locally stored directories?
> >>
> >> So, how does FS handle all the above?
> >> What does it pass to the recipient, for both external and internal calls?
> >> Is this part of FS configuration? IOW, do you have programmatic access to such information and can you specify exactly what is sent to an extension?
> >



Ken G i l l e t t

_/_/_/_/_/_/_/_/






From ivdreg at gmail.com  Tue Mar 16 11:58:17 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Tue, 16 Mar 2010 20:58:17 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
Message-ID: <b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>

Thanks Rupa,

This was good clarification. I want to use mod_limit via 4 servers but
because traffic is equally balanced between them I will use limit_hash and
calculate how many calls every server will serve.

One more question ? Is it necessary limit to transfer to limit_exceeded or
there is a way to set directly disconnect cause like in limit_hash ?

Cheers

2010/3/16 Rupa Schomaker <rupa at rupa.com>

> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
> to a database when using limit_hash.  If you want to use the database, then
> use plain limit.
>
> 2) If you want a max calls in a 24hrs period use you rate with limit_hash.
>  This should be about a true max per period, not concurrent.
>
> limit_hash realm id 50/86400
>
> should do what you want -- from memory.  double check the wiki/help to
> verify syntax.
>
> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Hi All,
>>
>> I have some questions regarding mod_limit.
>>
>> 1. I'm using 16674 in production but tested and with 17001. I try
>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>
>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>
>> and they are applied on inbound call profile in same extension.
>>
>>
>> 2. I need to implement rule maximum of 50 calls per day/24h from the same
>> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
>> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
>> but not calls processed as count. Any idea ?
>>
>> Thanks for any help
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Mar 16 12:10:58 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 16 Mar 2010 12:10:58 -0700
Subject: [Freeswitch-users] mod_cdr_csv
Message-ID: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>

What triggers mod_cdr_csv to log a call?  

I'm not sure if what I'm experiencing is due to my mistake or otherwise, but here's the situation.
Incoming call (via PSTN or a VoIP provider) -> Extension 105 and answered
105 transfers to 115, 115 answers, 105 completes transfer
The call progresses and is 115 finally hangs up.

The Master.csv only gets one line added to it - the 105 to 115 call.  Nothing about the incoming caller.

If I knew what triggered the log entry, I could try to track the issue down in my dial plan.

Thanks!
-Troy




From frank at carmickle.com  Tue Mar 16 12:12:13 2010
From: frank at carmickle.com (Frank Carmickle)
Date: Tue, 16 Mar 2010 15:12:13 -0400
Subject: [Freeswitch-users] call info display
In-Reply-To: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
	<94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
Message-ID: <20100316191212.GD21534@base.carmickle.com>

Hello

On Tue, Mar 16, Ken Gillett wrote:
> Previously I had a company with a call centre, in which we used a Meridian Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of DDI numbers. Each client was provided with one of those DDI numbers on which their customers would call us, so it was vital the operator knew which client's customer was calling. Within the configuration of the Option 11 I named each incoming DDI with the respective client's (Company) name and then that name was sent to the extension, along with the incoming CLI, so each operator knew which client's number had been called (and hence how to respond) and also who was calling, either their number, or name if it could be matched in an address directory (although cannot now remember if that was done by the switch itself or the extension phone).
> 
> Nothing strange about any of that, but I'm trying to relate those same requirements to the capabilities of FS. My assumption has been that FS would be able to match anything traditional telephone switches could manage, but I now need some confirmation of this.
> 
> So, are the callee ID and number the incoming line name and number respectively and it's those that FS sends to the extension for display?
> 
> Where does this data come from? Surely naming the incoming lines must be part of the configuration.
> 
> Can I name them to whatever I want?
> 
> Is the incoming CLI number sent directly to the extension (where it's up to the extension what it then displays)?
> Or
> Does FS do any matching to a name in a directory first and then send the matched name to the extension?
> 
> Sorry to keep asking - I'm sure this is very obvious to those who know, but I'm still trying to learn all about FS and what it can and can't do.
 
Most anything you want to do can be done.  You really need to set up a lab and get your hands dirty to find out what each scenario requires.  This is the only way you can really learn the ins and outs of FS.

--FC



From anthony.minessale at gmail.com  Tue Mar 16 12:16:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 14:16:46 -0500
Subject: [Freeswitch-users] call info display
In-Reply-To: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
	<94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
Message-ID: <191c3a031003161216m582eba60o58b1ab56999d589@mail.gmail.com>

Its automatically determined based on the info in the SIP traffic from the 2
respective ends of the call.
It's also configurable to be overridden from the dialplan.


On Tue, Mar 16, 2010 at 1:52 PM, Ken Gillett <ken at ukgb.net> wrote:

> Previously I had a company with a call centre, in which we used a Meridian
> Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of
> DDI numbers. Each client was provided with one of those DDI numbers on which
> their customers would call us, so it was vital the operator knew which
> client's customer was calling. Within the configuration of the Option 11 I
> named each incoming DDI with the respective client's (Company) name and then
> that name was sent to the extension, along with the incoming CLI, so each
> operator knew which client's number had been called (and hence how to
> respond) and also who was calling, either their number, or name if it could
> be matched in an address directory (although cannot now remember if that was
> done by the switch itself or the extension phone).
>
> Nothing strange about any of that, but I'm trying to relate those same
> requirements to the capabilities of FS. My assumption has been that FS would
> be able to match anything traditional telephone switches could manage, but I
> now need some confirmation of this.
>
> So, are the callee ID and number the incoming line name and number
> respectively and it's those that FS sends to the extension for display?
>
> Where does this data come from? Surely naming the incoming lines must be
> part of the configuration.
>
> Can I name them to whatever I want?
>
> Is the incoming CLI number sent directly to the extension (where it's up to
> the extension what it then displays)?
> Or
> Does FS do any matching to a name in a directory first and then send the
> matched name to the extension?
>
> Sorry to keep asking - I'm sure this is very obvious to those who know, but
> I'm still trying to learn all about FS and what it can and can't do.
>
>
> On 16 Mar 2010, at 17:11, Anthony Minessale wrote:
>
> > as i said it is phone specific.
> >
> > The callee id and caleeid number is sent
> >
> > on polycom and astra via sip UPDATE
> > on snom and FS->FS via INFO
> >
> >
> > There is no configuration necessary, its the default behavior.
> >
> >
> > On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:
> > Thanks, but could I ask for more specifics? Exactly what info is sent and
> is it sent as part of the SIP info, i.e. so any SIP device ought to display
> it correctly? Are there any docs I can read that will explain in detail
> about the information that is sent (i.e. call info, not SIP protocol stuff)
> and how to configureFS to send exactly what is required?
> >
> >
> > On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
> >
> > > depending on the phone FreeSWITCH will update the display any time a
> call is bridged.
> > >
> > > it works on polycom,snom and aastra and I am not sure if there are
> more.
> > >
> > >
> > > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
> > >> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
> > >>
> > >> When a call is received, the extension really needs to display
> information about the origin and the intended recipient of the call.
> Normally for an incoming external call, the former is supplied by the
> 'incoming CLI' (often linked to locally stored name) and the latter probably
> by associating a name with each incoming line, so in this way the recipient
> can see at a glance who called whom (or what).
> > >>
> > >> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
> > >>
> > >> So, how does FS handle all the above?
> > >> What does it pass to the recipient, for both external and internal
> calls?
> > >> Is this part of FS configuration? IOW, do you have programmatic access
> to such information and can you specify exactly what is sent to an
> extension?
> > >
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar 16 12:18:10 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 15:18:10 -0400
Subject: [Freeswitch-users] voicemail options
Message-ID: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>

I noticed that sendmail makes a segfault when FS is trying to send email
and sendmail has some emails locked in folder queue....

Regards

Franck
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From anthony.minessale at gmail.com  Tue Mar 16 12:29:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Mar 2010 14:29:31 -0500
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
Message-ID: <191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>

are you running FS as root or a regular user?
maybe there is a permission or resource limitation problem preventing it
from getting the larger stack space necessary?


On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

>  I noticed that sendmail makes a segfault when FS is trying to send email
> and sendmail has some emails locked in folder queue....
>
> Regards
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From robert.dyck at shaw.ca  Tue Mar 16 12:49:21 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Tue, 16 Mar 2010 12:49:21 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com>
Message-ID: <201003161249.22448.robert.dyck@shaw.ca>

F11, KDE4.4.1, pulseaudio

Is portaudio another sound server?

RD

On Tuesday 16 March 2010 11:29:32 am William Suffill wrote:
> FSComm interfaces with sound devices via portaudio at this point. If
> portaudio can't use your devices correctly that would cause it not to work
> in any portaudio application including FSComm.
> 
> What version of Fedora? I'm sure this is fixable but really depends more on
> the underlying system more than code in FSComm.
> 
> 
> -- W



From rupa at rupa.com  Tue Mar 16 12:41:17 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 14:41:17 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
Message-ID: <b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>

limit and limit_hash should work the same with regards to disconnect cause.
 If you see a difference, let me know.

On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Thanks Rupa,
>
> This was good clarification. I want to use mod_limit via 4 servers but
> because traffic is equally balanced between them I will use limit_hash and
> calculate how many calls every server will serve.
>
> One more question ? Is it necessary limit to transfer to limit_exceeded or
> there is a way to set directly disconnect cause like in limit_hash ?
>
> Cheers
>
> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>
> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
>> to a database when using limit_hash.  If you want to use the database, then
>> use plain limit.
>>
>> 2) If you want a max calls in a 24hrs period use you rate with limit_hash.
>>  This should be about a true max per period, not concurrent.
>>
>> limit_hash realm id 50/86400
>>
>> should do what you want -- from memory.  double check the wiki/help to
>> verify syntax.
>>
>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> I have some questions regarding mod_limit.
>>>
>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>
>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>
>>> and they are applied on inbound call profile in same extension.
>>>
>>>
>>> 2. I need to implement rule maximum of 50 calls per day/24h from the same
>>> CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in
>>> example "Limit calls to 5 calls every 10 minutes" is about concurrent call
>>> but not calls processed as count. Any idea ?
>>>
>>> Thanks for any help
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From devel at thom.fr.eu.org  Tue Mar 16 12:59:25 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Tue, 16 Mar 2010 20:59:25 +0100
Subject: [Freeswitch-users] voicemail options
In-Reply-To: <191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705>
	<191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
Message-ID: <006c01cac543$2e079510$8a16bf30$@fr.eu.org>

Are you running debian, I have the same problem on debian x64 (already
mentioned here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.h
tml) and cannot fix it.

I don?t know if it?s a priviledge problem, as I got the same results running
FS with or without ?u.

 

Fran?ois

 

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony
Minessale
Envoy? : mardi 16 mars 2010 20:30
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] voicemail options

 

are you running FS as root or a regular user?
maybe there is a permission or resource limitation problem preventing it
from getting the larger stack space necessary?



On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

I noticed that sendmail makes a segfault when FS is trying to send email

and sendmail has some emails locked in folder queue....

 

Regards

 

Franck


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
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pstn:+19193869900

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From freeswitch.org at todandlorna.com  Tue Mar 16 12:59:46 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Tue, 16 Mar 2010 13:59:46 -0600
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
Message-ID: <4B9FE332.7000809@todandlorna.com>

The way I understand it, it gets triggered by the a leg hanging up, same 
as xml_cdr.  I would imagine the channel changes, and if you were 
comparing the xml_cdr of the same call, you would see the original call 
as well.  You might also have some luck with looking at the b leg cdr, 
if you have that turned on.

That's the extent of my guesswork.

-Tod Hansmann

On 3/16/2010 1:10 PM, Troy Anderson wrote:
> What triggers mod_cdr_csv to log a call?
>
> I'm not sure if what I'm experiencing is due to my mistake or otherwise, but here's the situation.
> Incoming call (via PSTN or a VoIP provider) ->  Extension 105 and answered
> 105 transfers to 115, 115 answers, 105 completes transfer
> The call progresses and is 115 finally hangs up.
>
> The Master.csv only gets one line added to it - the 105 to 115 call.  Nothing about the incoming caller.
>
> If I knew what triggered the log entry, I could try to track the issue down in my dial plan.
>
> Thanks!
> -Troy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From infos at madovsky.org  Tue Mar 16 13:18:14 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:18:14 -0400
Subject: [Freeswitch-users] voicemail options
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705><191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
	<006c01cac543$2e079510$8a16bf30$@fr.eu.org>
Message-ID: <2549F4D1FBA54CEA9EAA7B0B370B6AB6@MOBILEE1705>

Hi,

no I'm on Fedora 10 64bits.
I'm going to check the source to find a way

Regards

  ----- Original Message ----- 
  From: devel at thom.fr.eu.org 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 3:59 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

  I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

   

  Fran?ois

   

   

  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
  Envoy? : mardi 16 mars 2010 20:30
  ? : freeswitch-users at lists.freeswitch.org
  Objet : Re: [Freeswitch-users] voicemail options

   

  are you running FS as root or a regular user?
  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

  I noticed that sendmail makes a segfault when FS is trying to send email

  and sendmail has some emails locked in folder queue....

   

  Regards

   

  Franck


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From ahmed.ajmal at gmail.com  Mon Mar 15 12:31:34 2010
From: ahmed.ajmal at gmail.com (Ahmed Bhaila)
Date: Tue, 16 Mar 2010 00:31:34 +0500
Subject: [Freeswitch-users] Fwd:  Set/Export channel variable
In-Reply-To: <9d22cc171003151228p405bfef3v70662dfc2cdc5914@mail.gmail.com>
References: <9d22cc171003151228p405bfef3v70662dfc2cdc5914@mail.gmail.com>
Message-ID: <9d22cc171003151231k209f47aodb09114431505496@mail.gmail.com>

I am looking at the FS Console and I have a ESL listener socket that is
hooked for HANGUP_COMPLETE event.


- ahmed

On Mon, Mar 15, 2010 at 6:50 AM, Ahmed Ajmal <ahmed.ajmal at
breezecom.ae <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>wrote:

>*  Hi
*>*
*>*
*>*
*>* I have now configured the gateways as suggested below but I am not able to
*>* see the sip_gateway_name variable when the call hangup event is complete. I
*>* am on Freeswitch 1.0.5.
*>*
*>*
*>*
*Where are you looking to find the variable?
-MC

From: freeswitch-users-bounces at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

[mailto:freeswitch-users-bounces at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] On
Behalf Of Brian
West
Sent: Wednesday, March 10, 2010 7:08 PM
To: freeswitch-users at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

Subject: Re: [Freeswitch-users] Set/Export channel variable



We already set you a variable called sip_gateway_name when a call is placed
out a gateway.  You're also doing this wrong you should NEVER EVER do

@$ipgwX as that data is in the gateway.  You should simply do
sofia/gateway/gwname/number



/b



On Mar 10, 2010, at 8:01 AM, Ahmed Ajmal wrote:





Hi



I am trying to set a channel variable on the bridge application using this:



<action application="bridge"
data="[gw=gateway1]sofia/gateway/gw/$number@$ipgw1 |
[gw=gateway2]sofia/gateway/gw/$number@$ipgw2"/>



The application runs fine, what I now need to do is set/export the 'gw'

channel variable for CDR so that I know which gateway was dialed. I have
tried using set and export application but that doesn't work. Please help.



Thanks

Ahmed
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From pavera at gmail.com  Mon Mar 15 23:59:38 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 00:59:38 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
Message-ID: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>

Hello,
Pretty new, but so far very happy with freeswitch!  The setup and config is
really nice, straightforward, and I appreciate the design of the system.
 I've worked with Asterisk pretty regularly for the last 4-5 years, and am
really looking for something more stable, and I think I've found it here!

One issue I've run into, and I'm not sure if its just that my use case is
abnormal, hence the email.

I've worked with PBX systems for a number of years (11 now...), and in all
of the systems I've worked with, call parking was kind of a "roaming"
feature.  By this I mean, you think party X (the intended recipient of the
call) is in the building, but they are not at their desk.  So, you park the
call, and then page, or by some other means attempt to locate party X and
tell them "get to the nearest phone and pick up 5901".  If you can't locate
party X or you can't find them in time, the call should time out back to the
parking party (receptionist) to be handled appropriately (message taken,
sent to voicemail, transferred to party X's boss)

This scenario requires a few things:
1) the receptionist/person who actually receives the call needs to be able
to put the call in a specific place (extension) that doesn't change
2) Party X (the intended recipient) needs to be able to access that specific
place (extension) from any phone inside the company
3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
inadequate, because if the receptionist parks 2 people on 5900, one for
party X and one for party Y, in that order, but party Y gets to a phone
first, they will be connected to party X's caller)
4) The receptionist has to be protected from or provided with sufficient
information to avoid connecting 2 customers with each other (IE the
valet_park in freeswitch is inadequate, because if she parks a caller for X
on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
what can she do?  What if she just forgets and sends that caller to 6001?
 Then the 2 customers are bridged together and thats a big no-no)  Also, I
haven't seen a way to make valet_park timeout?  But maybe I'm wrong
there...

The closest I can get to implementing the above scenario is by using
multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
receptionist phone to "notify" her that someone is already parked in a
certain queue.  Unfortunately, there is no enforcement of this policy by the
system, and if she fat fingers her transfer, she will double up a queue,
creating the problem indicated in 3 above.  The other shortcoming of this
setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
becomes impossible to appropriately handle call park.  If the receptionist
is away from her desk and attempts to answer a call, she can't tell which
parking spots are free at all on a regular 2 line phone.  Also, regular
users are prohibited from using call park as they cannot know which slots
are free (the need for this arises often in certain industries, such as law,
client is speaking with attorney, attorney finishes and says "let me connect
you with my assistant to schedule an appointment and handle x, y, and z for
you.", paralegal is not at desk, atty needs to park call and page his
assistant, maybe they are in the copy room, or the library doing
research...)

The asterisk parking lot seems to solve this problem quite elegantly
(although the stability issues of asterisk undermine it).  It is simple to
use (receptionist and everyone else only needs to remember 1 extension to
park any call), The callers/parking spots are maintained by the system to
prevent double parking, and each caller sits at a specific extension until
they are picked up or time out.

So, the question boils down to how do people using freeswitch view/use call
park?  Is there a "better" roaming feature that I just don't know about?  If
it is just something that hasn't been implemented/thought about this way
yet, any idea what it would take to get a call parking feature that
satisfies these requirements into freeswitch?

Thanks for your time, sorry for the long email, hopefully it explains well
enough what I'm trying to accomplish.

-Tom
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From jbrucehopkins at gmail.com  Tue Mar 16 07:41:48 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Tue, 16 Mar 2010 14:41:48 +0000
Subject: [Freeswitch-users] Speex transcoding question
In-Reply-To: <5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
References: <d03a10e01003160330x3bf7d7aemfa02a619273f6a26@mail.gmail.com>
	<d03a10e01003160336y1391bfe1hebebfa71d2caa1d1@mail.gmail.com>
	<d03a10e01003160426n6b2d52d6l25c68fb8541da862@mail.gmail.com>
	<5B407202-CFA6-4203-B39A-A0BE43A5B6B4@freeswitch.org>
Message-ID: <d03a10e01003160741l157dbb2fk437cd255042e7403@mail.gmail.com>

Sure.  I put the Freeswitch log in pastebin.  here are two Wireshark
captures in the condition where the call fails- run on two PC's running
Eyebeam clients.  Phone A initiates the call with only PCMA enabled.  Phone
B has only Speex enabled.

Thanks for taking the time to look at these.
Bruce

On 16 March 2010 13:53, Brian West <brian at freeswitch.org> wrote:

> Can you collect some logs and traces to show us this?
>
> /b
>
> On Mar 16, 2010, at 6:26 AM, Bruce Hopkins wrote:
>
> > OK - sorry to make such a meal of this.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From delorenzodesign at gmail.com  Tue Mar 16 11:06:46 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 14:06:46 -0400
Subject: [Freeswitch-users] Lua - playAndGetDigits
Message-ID: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>

I'm having some trouble getting consistent results with playAndGetDigits,
sometimes the digits are record and other times they're not.  It seemed to
have something to do with how quickly the digits were pressed, but now that
no longer seems to be the case.

I'm trying to have users press a confirmation code, in this case "1111"
although I'd like to make it accept "1" or "11" or "111" or "1111" if
possible.

Can anyone point me in the right direction?


    session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
.. provider .. "/1" .. number_to_call);

    while(session:ready()) do
       session:answer();
       session:setInputCallback("onInput","true");

       --playbackMessage(check_message);
       digits = session:playAndGetDigits(4, 4, 1, 5000, "",
get_digits_message, "", "[1]");
       freeswitch.consoleLog("info", digits .. "\n");
       freeswitch.consoleLog("info", "Did this check out? " ..
(check_message == true and "yes" or "no" .. "\n"));

       -- we got what we're looking for, so we can end this
       if(check_success) then
          freeswitch.consoleLog("info", "A positive response was received
from this check.\n");
          break;
       else
          -- not a positive response
          -- if we reached our max attempts, we're finished move on
          if(attempts == max_attempts) then
            freeswitch.consoleLog("info", "We've reached our maximum
attempts for this number.\n");
            break;
          end
       end
    end
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From pavera at gmail.com  Tue Mar 16 11:06:49 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 12:06:49 -0600
Subject: [Freeswitch-users] Call parking strategy/best practice in
	freeswitch?
Message-ID: <82dae1c01003161106n9f3766cga56623d8729331f6@mail.gmail.com>

Hello,
Pretty new, but so far very happy with freeswitch!  The setup and config is
really nice, straightforward, and I appreciate the design of the system.
 I've worked with Asterisk pretty regularly for the last 4-5 years, and am
really looking for something more stable, and I think I've found it here!

One issue I've run into, and I'm not sure if its just that my use case is
abnormal, hence the email.

I've worked with PBX systems for a number of years (11 now...), and in all
of the systems I've worked with, call parking was kind of a "roaming"
feature.  By this I mean, you think party X (the intended recipient of the
call) is in the building, but they are not at their desk.  So, you park the
call, and then page, or by some other means attempt to locate party X and
tell them "get to the nearest phone and pick up 5901".  If you can't locate
party X or you can't find them in time, the call should time out back to the
parking party (receptionist) to be handled appropriately (message taken,
sent to voicemail, transferred to party X's boss)

This scenario requires a few things:
1) the receptionist/person who actually receives the call needs to be able
to put the call in a specific place (extension) that doesn't change
2) Party X (the intended recipient) needs to be able to access that specific
place (extension) from any phone inside the company
3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
inadequate, because if the receptionist parks 2 people on 5900, one for
party X and one for party Y, in that order, but party Y gets to a phone
first, they will be connected to party X's caller)
4) The receptionist has to be protected from or provided with sufficient
information to avoid connecting 2 customers with each other (IE the
valet_park in freeswitch is inadequate, because if she parks a caller for X
on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
what can she do?  What if she just forgets and sends that caller to 6001?
 Then the 2 customers are bridged together and thats a big no-no)  Also, I
haven't seen a way to make valet_park timeout?  But maybe I'm wrong
there...

The closest I can get to implementing the above scenario is by using
multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
receptionist phone to "notify" her that someone is already parked in a
certain queue.  Unfortunately, there is no enforcement of this policy by the
system, and if she fat fingers her transfer, she will double up a queue,
creating the problem indicated in 3 above.  The other shortcoming of this
setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
becomes impossible to appropriately handle call park.  If the receptionist
is away from her desk and attempts to answer a call, she can't tell which
parking spots are free at all on a regular 2 line phone.  Also, regular
users are prohibited from using call park as they cannot know which slots
are free (the need for this arises often in certain industries, such as law,
client is speaking with attorney, attorney finishes and says "let me connect
you with my assistant to schedule an appointment and handle x, y, and z for
you.", paralegal is not at desk, atty needs to park call and page his
assistant, maybe they are in the copy room, or the library doing
research...)

The asterisk parking lot seems to solve this problem quite elegantly
(although the stability issues of asterisk undermine it).  It is simple to
use (receptionist and everyone else only needs to remember 1 extension to
park any call), The callers/parking spots are maintained by the system to
prevent double parking, and each caller sits at a specific extension until
they are picked up or time out.

So, the question boils down to how do people using freeswitch view/use call
park?  Is there a "better" roaming feature that I just don't know about?  If
it is just something that hasn't been implemented/thought about this way
yet, any idea what it would take to get a call parking feature that
satisfies these requirements into freeswitch?

Thanks for your time, sorry for the long email, hopefully it explains well
enough what I'm trying to accomplish.

-Tom
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From delorenzodesign at gmail.com  Tue Mar 16 12:08:56 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 15:08:56 -0400
Subject: [Freeswitch-users] Lua - playAndGetDigits
In-Reply-To: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>
References: <5c9dcbfb1003161106u55771b2lcc525a85b8c7968c@mail.gmail.com>
Message-ID: <5c9dcbfb1003161208n12ae669fw77a498d9517152d8@mail.gmail.com>

>
> I'm having some trouble getting consistent results with playAndGetDigits,
> sometimes the digits are record and other times they're not.  It seemed to
> have something to do with how quickly the digits were pressed, but now that
> no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case "1111"
> although I'd like to make it accept "1" or "11" or "111" or "1111" if
> possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
> .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
> get_digits_message, "", "[1]");
>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was received
> from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
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From delorenzodesign at gmail.com  Tue Mar 16 13:21:13 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 16:21:13 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
Message-ID: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>

I'm having some trouble getting consistent results with playAndGetDigits,
sometimes the digits are record and other times they're not.  It seemed to
have something to do with how quickly the digits were pressed, but now that
no longer seems to be the case.

I'm trying to have users press a confirmation code, in this case "1111"
although I'd like to make it accept "1" or "11" or "111" or "1111" if
possible.

Can anyone point me in the right direction?


    session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
.. provider .. "/1" .. number_to_call);

    while(session:ready()) do
       session:answer();
       session:setInputCallback("onInput","true");

       --playbackMessage(check_message);
       digits = session:playAndGetDigits(4, 4, 1, 5000, "",
get_digits_message, "", "[1]");
       freeswitch.consoleLog("info", digits .. "\n");
       freeswitch.consoleLog("info", "Did this check out? " ..
(check_message == true and "yes" or "no" .. "\n"));

       -- we got what we're looking for, so we can end this
       if(check_success) then
          freeswitch.consoleLog("info", "A positive response was received
from this check.\n");
          break;
       else
          -- not a positive response
          -- if we reached our max attempts, we're finished move on
          if(attempts == max_attempts) then
            freeswitch.consoleLog("info", "We've reached our maximum
attempts for this number.\n");
            break;
          end
       end
    end
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From brian at freeswitch.org  Tue Mar 16 13:26:03 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 15:26:03 -0500
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <4B9FE332.7000809@todandlorna.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
Message-ID: <5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>

By default its only logging A legs.  See config file <param name="legs" value="a"/> and change to <param name="legs" value="ab"/>

/b



On Mar 16, 2010, at 2:59 PM, Tod Hansmann wrote:

> The way I understand it, it gets triggered by the a leg hanging up, same 
> as xml_cdr.  I would imagine the channel changes, and if you were 
> comparing the xml_cdr of the same call, you would see the original call 
> as well.  You might also have some luck with looking at the b leg cdr, 
> if you have that turned on.
> 
> That's the extent of my guesswork.
> 
> -Tod Hansmann




From william.suffill at gmail.com  Tue Mar 16 13:27:10 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 16:27:10 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003161249.22448.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<6b65470d1003161129v792acf73g7c14e995af8c230b@mail.gmail.com> 
	<201003161249.22448.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>

PortAudio is a free, cross platform,
open-source<http://www.portaudio.com/license.html>,
audio I/O library.  It lets you write simple audio programs in 'C' that will
compile and run on many platforms
<http://www.portaudio.com/status.html>including
*Windows, Macintosh (8,9,X), Unix (OSS), SGI, and BeOS*. PortAudio is
intended to promote the exchange of audio synthesis software between
developers on different platforms.

PortAudio provides a very simple API for recording and/or playing sound
using a simple callback function.  Example programs are included that
synthesize sine waves and pink noise, perform fuzz distortion on a guitar,
list available audio devices, etc.
http://www.portaudio.com/

http://wiki.freeswitch.org/wiki/Mod_portaudio


Technically it is possible to get portaudio based applications to work with
pulseaudio but it seems to be an area of some difficulties now that
pulseaudio is used by default by newer distros.

I've had this same issue on my system w/ pulseaudio since it wants to
control all the sound devices itself. Killing pulseaudio should release the
devices and partaudio should detect them in that case. Not a long term fix I
know but still yet to spend enough time to discover the best alternative.

https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-January/003037.html
seems to indicate that it has been  an area of confusion for some time.

-- W
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From infos at madovsky.org  Tue Mar 16 13:35:15 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:35:15 -0400
Subject: [Freeswitch-users] voicemail options
References: <CAC3AB7291134C4DBC6B6E0791D3F193@MOBILEE1705><191c3a031003161229p638c3d92g5da15435a9727853@mail.gmail.com>
	<006c01cac543$2e079510$8a16bf30$@fr.eu.org>
Message-ID: <D4EC8B7224F340BEAA2D439BC95DF5A3@MOBILEE1705>

ok, I tried 50 times to leave a message,
it worked only one time.
It seems that sendmail needs to be free to send the email....

  ----- Original Message ----- 
  From: devel at thom.fr.eu.org 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 3:59 PM
  Subject: Re: [Freeswitch-users] voicemail options


  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

  I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

   

  Fran?ois

   

   

  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
  Envoy? : mardi 16 mars 2010 20:30
  ? : freeswitch-users at lists.freeswitch.org
  Objet : Re: [Freeswitch-users] voicemail options

   

  are you running FS as root or a regular user?
  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

  I noticed that sendmail makes a segfault when FS is trying to send email

  and sendmail has some emails locked in folder queue....

   

  Regards

   

  Franck


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Tue Mar 16 13:42:19 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 16 Mar 2010 16:42:19 -0400
Subject: [Freeswitch-users] ivr call and secret keycode
Message-ID: <B6EAD9F6B34D4B5095009C8C2A1ECF62@MOBILEE1705>

Hi all,

is anyone already worked on an ivr that make a call so
the callee needs to enter a keycode to validate a form filled before on a website for example ?

Thanks

Franck
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From troy at tlainvestments.com  Tue Mar 16 13:59:58 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 16 Mar 2010 13:59:58 -0700
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
Message-ID: <86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>

Thanks, guys!  Setting the legs param to "ab" brought in lots more cdr data, including the LOSE_RACE legs.  I'm still a bit confused, though - can you explain what is considered a-leg vs. b-leg in this context?  I would have thought the call from outside into the system would constitute an a-leg.

Thanks!
Troy

On Mar 16, 2010, at 1:26 PM, Brian West wrote:

> By default its only logging A legs.  See config file <param name="legs" value="a"/> and change to <param name="legs" value="ab"/>
> 
> /b
> 
> 
> 
> On Mar 16, 2010, at 2:59 PM, Tod Hansmann wrote:
> 
>> The way I understand it, it gets triggered by the a leg hanging up, same 
>> as xml_cdr.  I would imagine the channel changes, and if you were 
>> comparing the xml_cdr of the same call, you would see the original call 
>> as well.  You might also have some luck with looking at the b leg cdr, 
>> if you have that turned on.
>> 
>> That's the extent of my guesswork.
>> 
>> -Tod Hansmann
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Tue Mar 16 14:00:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:00:12 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
Message-ID: <87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>

On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> I'm having some trouble getting consistent results with playAndGetDigits,
> sometimes the digits are record and other times they're not.  It seemed to
> have something to do with how quickly the digits were pressed, but now that
> no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case "1111"
> although I'd like to make it accept "1" or "11" or "111" or "1111" if
> possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
> .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
> get_digits_message, "", "[1]");
>

You are specifying a minimum of four and a maximum of four digits. Also,
your regex will match any string that has a digit 1 in it, which may or may
not be what you are looking for. Lastly, you haven't specified a terminator
key (like "#") but you are giving the user only one try and waiting 5000ms
for the entry. Try something like this:

digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
"", "\\d+");

Let us know if that works. Oh, one other thing: you might want to add an
invalid message because playAndGetDigits will handle invalid input for you.
-MC

P.S. - This topic is covered in greater detail in Chapter 7 of the upcoming
FreeSWITCH book. ;)



>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was received
> from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
>
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From robert.dyck at shaw.ca  Tue Mar 16 14:11:40 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Tue, 16 Mar 2010 14:11:40 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca>
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
Message-ID: <201003161411.43845.robert.dyck@shaw.ca>

Killing pulse audio didn't help. Is it possible to get debugging info from 
fscomm?



From msc at freeswitch.org  Tue Mar 16 14:14:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:14:17 -0700
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
Message-ID: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>

On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com> wrote:

> Hello,
> Pretty new, but so far very happy with freeswitch!  The setup and config is
> really nice, straightforward, and I appreciate the design of the system.
>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
> really looking for something more stable, and I think I've found it here!
>
> One issue I've run into, and I'm not sure if its just that my use case is
> abnormal, hence the email.
>
> I've worked with PBX systems for a number of years (11 now...), and in all
> of the systems I've worked with, call parking was kind of a "roaming"
> feature.  By this I mean, you think party X (the intended recipient of the
> call) is in the building, but they are not at their desk.  So, you park the
> call, and then page, or by some other means attempt to locate party X and
> tell them "get to the nearest phone and pick up 5901".  If you can't locate
> party X or you can't find them in time, the call should time out back to the
> parking party (receptionist) to be handled appropriately (message taken,
> sent to voicemail, transferred to party X's boss)
>
> This scenario requires a few things:
> 1) the receptionist/person who actually receives the call needs to be able
> to put the call in a specific place (extension) that doesn't change
> 2) Party X (the intended recipient) needs to be able to access that
> specific place (extension) from any phone inside the company
> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
> inadequate, because if the receptionist parks 2 people on 5900, one for
> party X and one for party Y, in that order, but party Y gets to a phone
> first, they will be connected to party X's caller)
> 4) The receptionist has to be protected from or provided with sufficient
> information to avoid connecting 2 customers with each other (IE the
> valet_park in freeswitch is inadequate, because if she parks a caller for X
> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
> what can she do?  What if she just forgets and sends that caller to 6001?
>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
> there...
>
> The closest I can get to implementing the above scenario is by using
> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
> receptionist phone to "notify" her that someone is already parked in a
> certain queue.  Unfortunately, there is no enforcement of this policy by the
> system, and if she fat fingers her transfer, she will double up a queue,
> creating the problem indicated in 3 above.  The other shortcoming of this
> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
> becomes impossible to appropriately handle call park.  If the receptionist
> is away from her desk and attempts to answer a call, she can't tell which
> parking spots are free at all on a regular 2 line phone.  Also, regular
> users are prohibited from using call park as they cannot know which slots
> are free (the need for this arises often in certain industries, such as law,
> client is speaking with attorney, attorney finishes and says "let me connect
> you with my assistant to schedule an appointment and handle x, y, and z for
> you.", paralegal is not at desk, atty needs to park call and page his
> assistant, maybe they are in the copy room, or the library doing
> research...)
>
> The asterisk parking lot seems to solve this problem quite elegantly
> (although the stability issues of asterisk undermine it).  It is simple to
> use (receptionist and everyone else only needs to remember 1 extension to
> park any call), The callers/parking spots are maintained by the system to
> prevent double parking, and each caller sits at a specific extension until
> they are picked up or time out.
>
> So, the question boils down to how do people using freeswitch view/use call
> park?  Is there a "better" roaming feature that I just don't know about?  If
> it is just something that hasn't been implemented/thought about this way
> yet, any idea what it would take to get a call parking feature that
> satisfies these requirements into freeswitch?
>
> Thanks for your time, sorry for the long email, hopefully it explains well
> enough what I'm trying to accomplish.
>
> -Tom
>

Tom,

First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
thank you for actually looking at the documentation before asking your
question.

You've got yourself quite the little conundrum. Many of us use park in a
more limited role, or we rely on the receptionist not to do silly things
like sending one external caller to the slot of another external caller. We
all agree that's a bad idea, unless it is done deliberately to connect two
external parties. It sounds to me like this particular situation needs an
operator panel app where the receptionist can have a pretty web page showing
what calls are parked where. To my knowledge, this does not yet exist in
FreeSWITCH. The good news is that FS provides all the requisite building
blocks to create such an application using the event socket. How do you feel
about doing a little development? :)

-MC
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From msc at freeswitch.org  Tue Mar 16 14:19:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 14:19:33 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
Message-ID: <87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>

On Mon, Mar 15, 2010 at 1:21 PM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Hello.
>
> How can I intercept a fifo? I tried to use the example of intecept
> default dialplan, but in no fifo dialed_extension.
>

Are you trying to intercept the call that is ringing at a FIFO agent's
phone?
-MC


>
>    <extension name="operators">
>      <condition field="destination_number" expression="^operators$">
>            <action application="bind_meta_app" data="2 b s
>
> record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>            <action application="hash"
>
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>            <action application="hash"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>            <action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@
> ${domain_name}
> var callgroup)}"/>
>            <action application="hash"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>            <action application="answer"/>
>            <action application="playback"
> data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
>            <action application="playback"
> data="ivr/ivr-hold_connect_call.wav"/>
>            <action application="set" data="fifo_music=$${hold_music}"/>
>            <action application="fifo" data="operators@$${domain} in
> undef"/>
>      </condition>
>    </extension>
>
>    <fifo name="operators@$${domain}" importance="0">
>      <member timeout="15" simo="1"
> lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@
> $${domain}</member>
>    </fifo>
>  </fifos>
> --
> Best regards, Vladimir Elizarov
>
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From brian at freeswitch.org  Tue Mar 16 14:10:00 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Mar 2010 16:10:00 -0500
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
	<86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
Message-ID: <3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>

A is ingress to the switch.. B is egress from the switch... remember each bridged call consists of two legs.  But their are cases that can blur those lines.

/b

On Mar 16, 2010, at 3:59 PM, Troy Anderson wrote:

> Thanks, guys!  Setting the legs param to "ab" brought in lots more cdr data, including the LOSE_RACE legs.  I'm still a bit confused, though - can you explain what is considered a-leg vs. b-leg in this context?  I would have thought the call from outside into the system would constitute an a-leg.
> 
> Thanks!
> Troy




From william.suffill at gmail.com  Tue Mar 16 14:36:28 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 16 Mar 2010 17:36:28 -0400
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003161411.43845.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca> 
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com> 
	<201003161411.43845.robert.dyck@shaw.ca>
Message-ID: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>

With FSComm running you can use fs_cli to try pa list. If u can't list the
devices that would be the problem. I'm going to try to dig into this further
on 1 of my desktops w/ pulseaudio.

-- W
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From msc at freeswitch.org  Tue Mar 16 16:06:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 16 Mar 2010 16:06:49 -0700
Subject: [Freeswitch-users] mod_cdr_csv
In-Reply-To: <3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>
References: <AF87C42F-A82E-4A23-B26F-B6F1ED1FE1C6@tlainvestments.com>
	<4B9FE332.7000809@todandlorna.com>
	<5D7CE6D6-E059-4DEE-921A-ACBBB7C6AF2C@freeswitch.org>
	<86AAB240-4227-4F85-8F2E-910C36685833@tlainvestments.com>
	<3F3E6446-7E6B-4CBD-9BA0-D85323033B76@freeswitch.org>
Message-ID: <87f2f3b91003161606s1c1a3996t65a9ddbbd464e010@mail.gmail.com>

On Tue, Mar 16, 2010 at 2:10 PM, Brian West <brian at freeswitch.org> wrote:

> A is ingress to the switch.. B is egress from the switch... remember each
> bridged call consists of two legs.  But their are cases that can blur those
> lines.
>
> FYI, this is a good piece of information. I added it explicitly to the call
legs wiki page that we just started:
http://wiki.freeswitch.org/wiki/Call_Legs

If anyone else has something to add by all means do so.
-MC
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From jmesquita at freeswitch.org  Tue Mar 16 16:13:26 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 16 Mar 2010 20:13:26 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161249.22448.robert.dyck@shaw.ca>
	<6b65470d1003161327t463ff0e9j96dc2373f3cc49ac@mail.gmail.com>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
Message-ID: <ebbc2d551003161613g15bfc624h112d446ee68f1716@mail.gmail.com>

Thanks for the help W!

Jo?o Mesquita


On Tue, Mar 16, 2010 at 6:36 PM, William Suffill
<william.suffill at gmail.com>wrote:

> With FSComm running you can use fs_cli to try pa list. If u can't list the
> devices that would be the problem. I'm going to try to dig into this further
> on 1 of my desktops w/ pulseaudio.
>
> -- W
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 16:55:56 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 19:55:56 -0400
Subject: [Freeswitch-users] MWI Question
Message-ID: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>

Hi All,

I delete all the voice mails in the voice mail folder for a particular
extension.

But the phone[AASTRA 9133i]  MWI keep on blinking and and display showing
the no of messages.

How can I fix this issue.

Thanks
Lloyd
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From delorenzodesign at gmail.com  Tue Mar 16 17:23:27 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 20:23:27 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
Message-ID: <5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>

So I tried Michael's suggestion of:

digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
"", "\\d+");

but it still didn't detect that I had pressed any keys.  What would cause
this?  Is there someplace I can check to see if anything at all is being
detected?


On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> I'm having some trouble getting consistent results with playAndGetDigits,
>> sometimes the digits are record and other times they're not.  It seemed to
>> have something to do with how quickly the digits were pressed, but now that
>> no longer seems to be the case.
>>
>> I'm trying to have users press a confirmation code, in this case "1111"
>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>> possible.
>>
>> Can anyone point me in the right direction?
>>
>>
>>     session = freeswitch.Session("{ignore_early_media=true}sofia/gateway/"
>> .. provider .. "/1" .. number_to_call);
>>
>>     while(session:ready()) do
>>        session:answer();
>>        session:setInputCallback("onInput","true");
>>
>>        --playbackMessage(check_message);
>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>> get_digits_message, "", "[1]");
>>
>
> You are specifying a minimum of four and a maximum of four digits. Also,
> your regex will match any string that has a digit 1 in it, which may or may
> not be what you are looking for. Lastly, you haven't specified a terminator
> key (like "#") but you are giving the user only one try and waiting 5000ms
> for the entry. Try something like this:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
> "", "\\d+");
>
> Let us know if that works. Oh, one other thing: you might want to add an
> invalid message because playAndGetDigits will handle invalid input for you.
> -MC
>
> P.S. - This topic is covered in greater detail in Chapter 7 of the upcoming
> FreeSWITCH book. ;)
>
>
>
>>        freeswitch.consoleLog("info", digits .. "\n");
>>        freeswitch.consoleLog("info", "Did this check out? " ..
>> (check_message == true and "yes" or "no" .. "\n"));
>>
>>        -- we got what we're looking for, so we can end this
>>        if(check_success) then
>>           freeswitch.consoleLog("info", "A positive response was received
>> from this check.\n");
>>           break;
>>        else
>>           -- not a positive response
>>           -- if we reached our max attempts, we're finished move on
>>           if(attempts == max_attempts) then
>>             freeswitch.consoleLog("info", "We've reached our maximum
>> attempts for this number.\n");
>>             break;
>>           end
>>        end
>>     end
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 17:37:38 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 20:37:38 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
Message-ID: <8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>

set the console log level to 7. Then you can see the DTMF values in console
log , while you are pressing the keys.

Lloyd

On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> So I tried Michael's suggestion of:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
> "", "\\d+");
>
> but it still didn't detect that I had pressed any keys.  What would cause
> this?  Is there someplace I can check to see if anything at all is being
> detected?
>
>
> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> I'm having some trouble getting consistent results with playAndGetDigits,
>>> sometimes the digits are record and other times they're not.  It seemed to
>>> have something to do with how quickly the digits were pressed, but now that
>>> no longer seems to be the case.
>>>
>>> I'm trying to have users press a confirmation code, in this case "1111"
>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>> possible.
>>>
>>> Can anyone point me in the right direction?
>>>
>>>
>>>     session =
>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>> "/1" .. number_to_call);
>>>
>>>     while(session:ready()) do
>>>        session:answer();
>>>        session:setInputCallback("onInput","true");
>>>
>>>        --playbackMessage(check_message);
>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>> get_digits_message, "", "[1]");
>>>
>>
>> You are specifying a minimum of four and a maximum of four digits. Also,
>> your regex will match any string that has a digit 1 in it, which may or may
>> not be what you are looking for. Lastly, you haven't specified a terminator
>> key (like "#") but you are giving the user only one try and waiting 5000ms
>> for the entry. Try something like this:
>>
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>> "", "\\d+");
>>
>> Let us know if that works. Oh, one other thing: you might want to add an
>> invalid message because playAndGetDigits will handle invalid input for you.
>> -MC
>>
>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>> upcoming FreeSWITCH book. ;)
>>
>>
>>
>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>> (check_message == true and "yes" or "no" .. "\n"));
>>>
>>>        -- we got what we're looking for, so we can end this
>>>        if(check_success) then
>>>           freeswitch.consoleLog("info", "A positive response was received
>>> from this check.\n");
>>>           break;
>>>        else
>>>           -- not a positive response
>>>           -- if we reached our max attempts, we're finished move on
>>>           if(attempts == max_attempts) then
>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>> attempts for this number.\n");
>>>             break;
>>>           end
>>>        end
>>>     end
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 17:43:31 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 20:43:31 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
Message-ID: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>

when a user parked to a slot how to handle the park time out. Is there any
way to transfer to the call back to a receptionist.

Thanks
Lloyd



On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>
>> Hello,
>> Pretty new, but so far very happy with freeswitch!  The setup and config
>> is really nice, straightforward, and I appreciate the design of the system.
>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>> really looking for something more stable, and I think I've found it here!
>>
>> One issue I've run into, and I'm not sure if its just that my use case is
>> abnormal, hence the email.
>>
>> I've worked with PBX systems for a number of years (11 now...), and in all
>> of the systems I've worked with, call parking was kind of a "roaming"
>> feature.  By this I mean, you think party X (the intended recipient of the
>> call) is in the building, but they are not at their desk.  So, you park the
>> call, and then page, or by some other means attempt to locate party X and
>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>> party X or you can't find them in time, the call should time out back to the
>> parking party (receptionist) to be handled appropriately (message taken,
>> sent to voicemail, transferred to party X's boss)
>>
>> This scenario requires a few things:
>> 1) the receptionist/person who actually receives the call needs to be able
>> to put the call in a specific place (extension) that doesn't change
>> 2) Party X (the intended recipient) needs to be able to access that
>> specific place (extension) from any phone inside the company
>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>> inadequate, because if the receptionist parks 2 people on 5900, one for
>> party X and one for party Y, in that order, but party Y gets to a phone
>> first, they will be connected to party X's caller)
>> 4) The receptionist has to be protected from or provided with sufficient
>> information to avoid connecting 2 customers with each other (IE the
>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>> what can she do?  What if she just forgets and sends that caller to 6001?
>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>> there...
>>
>> The closest I can get to implementing the above scenario is by using
>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>> receptionist phone to "notify" her that someone is already parked in a
>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>> system, and if she fat fingers her transfer, she will double up a queue,
>> creating the problem indicated in 3 above.  The other shortcoming of this
>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>> becomes impossible to appropriately handle call park.  If the receptionist
>> is away from her desk and attempts to answer a call, she can't tell which
>> parking spots are free at all on a regular 2 line phone.  Also, regular
>> users are prohibited from using call park as they cannot know which slots
>> are free (the need for this arises often in certain industries, such as law,
>> client is speaking with attorney, attorney finishes and says "let me connect
>> you with my assistant to schedule an appointment and handle x, y, and z for
>> you.", paralegal is not at desk, atty needs to park call and page his
>> assistant, maybe they are in the copy room, or the library doing
>> research...)
>>
>> The asterisk parking lot seems to solve this problem quite elegantly
>> (although the stability issues of asterisk undermine it).  It is simple to
>> use (receptionist and everyone else only needs to remember 1 extension to
>> park any call), The callers/parking spots are maintained by the system to
>> prevent double parking, and each caller sits at a specific extension until
>> they are picked up or time out.
>>
>> So, the question boils down to how do people using freeswitch view/use
>> call park?  Is there a "better" roaming feature that I just don't know
>> about?  If it is just something that hasn't been implemented/thought about
>> this way yet, any idea what it would take to get a call parking feature that
>> satisfies these requirements into freeswitch?
>>
>> Thanks for your time, sorry for the long email, hopefully it explains well
>> enough what I'm trying to accomplish.
>>
>> -Tom
>>
>
> Tom,
>
> First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
> thank you for actually looking at the documentation before asking your
> question.
>
> You've got yourself quite the little conundrum. Many of us use park in a
> more limited role, or we rely on the receptionist not to do silly things
> like sending one external caller to the slot of another external caller. We
> all agree that's a bad idea, unless it is done deliberately to connect two
> external parties. It sounds to me like this particular situation needs an
> operator panel app where the receptionist can have a pretty web page showing
> what calls are parked where. To my knowledge, this does not yet exist in
> FreeSWITCH. The good news is that FS provides all the requisite building
> blocks to create such an application using the event socket. How do you feel
> about doing a little development? :)
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From delorenzodesign at gmail.com  Tue Mar 16 17:51:38 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 20:51:38 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
Message-ID: <5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>

Ok, I updated the console log level, and there are no DTMF values being
detected:

4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
160 bytes per 20ms
2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
sofia/external/19736327407!
2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
sofia/external/19736327407 entering state [completing][200]
2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
v=0
o=root 21648 21649 IN IP4 74.51.38.156
s=session
c=IN IP4 74.51.38.156
t=0 0
m=audio 15150 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
sofia/external/19736327407 entering state [ready][200]
2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
[sofia/external/19736327407] has been answered
2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
Resulted in Success: [sofia/external/19736327407]
2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
(sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
CS_SOFT_EXECUTE
2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
sofia/external/19736327407 Standard SOFT_EXECUTE
2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE going to sleep
2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
confirmed.
2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
file
2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received during
call from playAndGetDigits:
2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out? no

On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> set the console log level to 7. Then you can see the DTMF values in console
> log , while you are pressing the keys.
>
> Lloyd
>
>
> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> So I tried Michael's suggestion of:
>>
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>> "", "\\d+");
>>
>> but it still didn't detect that I had pressed any keys.  What would cause
>> this?  Is there someplace I can check to see if anything at all is being
>> detected?
>>
>>
>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> I'm having some trouble getting consistent results with
>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>> not.  It seemed to have something to do with how quickly the digits were
>>>> pressed, but now that no longer seems to be the case.
>>>>
>>>> I'm trying to have users press a confirmation code, in this case "1111"
>>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>> possible.
>>>>
>>>> Can anyone point me in the right direction?
>>>>
>>>>
>>>>     session =
>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>> "/1" .. number_to_call);
>>>>
>>>>     while(session:ready()) do
>>>>        session:answer();
>>>>        session:setInputCallback("onInput","true");
>>>>
>>>>        --playbackMessage(check_message);
>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>> get_digits_message, "", "[1]");
>>>>
>>>
>>> You are specifying a minimum of four and a maximum of four digits. Also,
>>> your regex will match any string that has a digit 1 in it, which may or may
>>> not be what you are looking for. Lastly, you haven't specified a terminator
>>> key (like "#") but you are giving the user only one try and waiting 5000ms
>>> for the entry. Try something like this:
>>>
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>>> "", "\\d+");
>>>
>>> Let us know if that works. Oh, one other thing: you might want to add an
>>> invalid message because playAndGetDigits will handle invalid input for you.
>>> -MC
>>>
>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>> upcoming FreeSWITCH book. ;)
>>>
>>>
>>>
>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>
>>>>        -- we got what we're looking for, so we can end this
>>>>        if(check_success) then
>>>>           freeswitch.consoleLog("info", "A positive response was
>>>> received from this check.\n");
>>>>           break;
>>>>        else
>>>>           -- not a positive response
>>>>           -- if we reached our max attempts, we're finished move on
>>>>           if(attempts == max_attempts) then
>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>> attempts for this number.\n");
>>>>             break;
>>>>           end
>>>>        end
>>>>     end
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 18:07:15 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 21:07:15 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
Message-ID: <8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>

here is the test script I used

session:answer();
digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
"", "\\d+");

Here is the console log

2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
*2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800*
*2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640*
*2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800*
*2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640*
2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
[1234][\d+]

You may have some issue with the provider. Check using a local extension.


On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> Ok, I updated the console log level, and there are no DTMF values being
> detected:
>
> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
> 160 bytes per 20ms
> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
> sofia/external/19736327407!
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
> sofia/external/19736327407 entering state [completing][200]
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
> v=0
> o=root 21648 21649 IN IP4 74.51.38.156
> s=session
> c=IN IP4 74.51.38.156
> t=0 0
> m=audio 15150 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
> sofia/external/19736327407 entering state [ready][200]
> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
> [sofia/external/19736327407] has been answered
> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
> Resulted in Success: [sofia/external/19736327407]
> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
> CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
> sofia/external/19736327407 [BREAK]
> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
> (sofia/external/19736327407) State SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
> sofia/external/19736327407 Standard SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
> confirmed.
> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
> file
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
> during call from playAndGetDigits:
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
> no
>
>
> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> set the console log level to 7. Then you can see the DTMF values in
>> console log , while you are pressing the keys.
>>
>> Lloyd
>>
>>
>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> So I tried Michael's suggestion of:
>>>
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", get_digits_message,
>>> "", "\\d+");
>>>
>>> but it still didn't detect that I had pressed any keys.  What would cause
>>> this?  Is there someplace I can check to see if anything at all is being
>>> detected?
>>>
>>>
>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> I'm having some trouble getting consistent results with
>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>> pressed, but now that no longer seems to be the case.
>>>>>
>>>>> I'm trying to have users press a confirmation code, in this case "1111"
>>>>> although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>> possible.
>>>>>
>>>>> Can anyone point me in the right direction?
>>>>>
>>>>>
>>>>>     session =
>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>> "/1" .. number_to_call);
>>>>>
>>>>>     while(session:ready()) do
>>>>>        session:answer();
>>>>>        session:setInputCallback("onInput","true");
>>>>>
>>>>>        --playbackMessage(check_message);
>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>> get_digits_message, "", "[1]");
>>>>>
>>>>
>>>> You are specifying a minimum of four and a maximum of four digits. Also,
>>>> your regex will match any string that has a digit 1 in it, which may or may
>>>> not be what you are looking for. Lastly, you haven't specified a terminator
>>>> key (like "#") but you are giving the user only one try and waiting 5000ms
>>>> for the entry. Try something like this:
>>>>
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> get_digits_message, "", "\\d+");
>>>>
>>>> Let us know if that works. Oh, one other thing: you might want to add an
>>>> invalid message because playAndGetDigits will handle invalid input for you.
>>>> -MC
>>>>
>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>> upcoming FreeSWITCH book. ;)
>>>>
>>>>
>>>>
>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>
>>>>>        -- we got what we're looking for, so we can end this
>>>>>        if(check_success) then
>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>> received from this check.\n");
>>>>>           break;
>>>>>        else
>>>>>           -- not a positive response
>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>           if(attempts == max_attempts) then
>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>> attempts for this number.\n");
>>>>>             break;
>>>>>           end
>>>>>        end
>>>>>     end
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Tue Mar 16 18:08:32 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 20:08:32 -0500
Subject: [Freeswitch-users] MWI Question
In-Reply-To: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>
References: <8a19bf2e1003161655n1a6be912jbca20bef5c82a6e9@mail.gmail.com>
Message-ID: <b39273531003161808y6cf3fe8aqda1d68421cf33c2f@mail.gmail.com>

Next time use vm_delete rather than removing the files.  This may still work
for you.

"vm_delete id at domain" should work in your case.

If it doesn't, connect to the voicemail database and issue:

delete from voicemail_msgs where username='id' and domain='domain';

On Tue, Mar 16, 2010 at 6:55 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I delete all the voice mails in the voice mail folder for a particular
> extension.
>
> But the phone[AASTRA 9133i]  MWI keep on blinking and and display showing
> the no of messages.
>
> How can I fix this issue.
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Tue Mar 16 18:10:29 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 16 Mar 2010 20:10:29 -0500
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
Message-ID: <b39273531003161810m521cc9a0mba8aa9de103fcf03@mail.gmail.com>

I would think you could use:

sched_transfer,Schedule a transfer in the future,[+]<time> <extension>
<dialplan> <context>,mod_dptools

before transfering to the slot.  I haven't tried it, but don't see why it
wouldn't work.  More docs on the wiki.

On Tue, Mar 16, 2010 at 7:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> when a user parked to a slot how to handle the park time out. Is there any
> way to transfer to the call back to a receptionist.
>
> Thanks
> Lloyd
>
>
>
> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From delorenzodesign at gmail.com  Tue Mar 16 18:16:10 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 21:16:10 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
Message-ID: <5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>

What do you mean about the provider?  I'm not sure I understand that part.
I'm trying to call a number and have the call's recipient type the digits
in.

On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> here is the test script I used
>
> session:answer();
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
> "", "\\d+");
>
> Here is the console log
>
> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
> file
> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800*
> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640*
> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800*
> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640*
> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
> [1234][\d+]
>
> You may have some issue with the provider. Check using a local extension.
>
>
> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> Ok, I updated the console log level, and there are no DTMF values being
>> detected:
>>
>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer [soft]
>> 160 bytes per 20ms
>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>> sofia/external/19736327407!
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>> sofia/external/19736327407 entering state [completing][200]
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>> v=0
>> o=root 21648 21649 IN IP4 74.51.38.156
>> s=session
>> c=IN IP4 74.51.38.156
>> t=0 0
>> m=audio 15150 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>>
>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>> sofia/external/19736327407 entering state [ready][200]
>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>> [sofia/external/19736327407] has been answered
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>> Resulted in Success: [sofia/external/19736327407]
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>> CS_SOFT_EXECUTE
>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
>> sofia/external/19736327407 [BREAK]
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>> (sofia/external/19736327407) State SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>> sofia/external/19736327407 Standard SOFT_EXECUTE
>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>> confirmed.
>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done playing
>> file
>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>> during call from playAndGetDigits:
>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
>> no
>>
>>
>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> set the console log level to 7. Then you can see the DTMF values in
>>> console log , while you are pressing the keys.
>>>
>>> Lloyd
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> So I tried Michael's suggestion of:
>>>>
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> get_digits_message, "", "\\d+");
>>>>
>>>> but it still didn't detect that I had pressed any keys.  What would
>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>> being detected?
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> I'm having some trouble getting consistent results with
>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>
>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>> possible.
>>>>>>
>>>>>> Can anyone point me in the right direction?
>>>>>>
>>>>>>
>>>>>>     session =
>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>> "/1" .. number_to_call);
>>>>>>
>>>>>>     while(session:ready()) do
>>>>>>        session:answer();
>>>>>>        session:setInputCallback("onInput","true");
>>>>>>
>>>>>>        --playbackMessage(check_message);
>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>> get_digits_message, "", "[1]");
>>>>>>
>>>>>
>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> get_digits_message, "", "\\d+");
>>>>>
>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>> you.
>>>>> -MC
>>>>>
>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>> upcoming FreeSWITCH book. ;)
>>>>>
>>>>>
>>>>>
>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>
>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>        if(check_success) then
>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>> received from this check.\n");
>>>>>>           break;
>>>>>>        else
>>>>>>           -- not a positive response
>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>           if(attempts == max_attempts) then
>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>> attempts for this number.\n");
>>>>>>             break;
>>>>>>           end
>>>>>>        end
>>>>>>     end
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Tue Mar 16 18:34:46 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 16 Mar 2010 21:34:46 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
Message-ID: <8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>

You inbound DID Provider


On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> What do you mean about the provider?  I'm not sure I understand that part.
> I'm trying to call a number and have the call's recipient type the digits
> in.
>
>
> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> here is the test script I used
>>
>> session:answer();
>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
>> "", "\\d+");
>>
>> Here is the console log
>>
>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done playing
>> file
>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 1:800
>> *
>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 2:640
>> *
>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 3:800
>> *
>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF 4:640
>> *
>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>> [1234][\d+]
>>
>> You may have some issue with the provider. Check using a local extension.
>>
>>
>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>>> Ok, I updated the console log level, and there are no DTMF values being
>>> detected:
>>>
>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>> sofia/external/19736327407!
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>> sofia/external/19736327407 entering state [completing][200]
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>> v=0
>>> o=root 21648 21649 IN IP4 74.51.38.156
>>> s=session
>>> c=IN IP4 74.51.38.156
>>> t=0 0
>>> m=audio 15150 RTP/AVP 0 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>>
>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>> sofia/external/19736327407 entering state [ready][200]
>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>> [sofia/external/19736327407] has been answered
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>>> Resulted in Success: [sofia/external/19736327407]
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>> CS_SOFT_EXECUTE
>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send signal
>>> sofia/external/19736327407 [BREAK]
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>> confirmed.
>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>> playing file
>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>> during call from playAndGetDigits:
>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check out?
>>> no
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> set the console log level to 7. Then you can see the DTMF values in
>>>> console log , while you are pressing the keys.
>>>>
>>>> Lloyd
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> So I tried Michael's suggestion of:
>>>>>
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> get_digits_message, "", "\\d+");
>>>>>
>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>> being detected?
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>
>>>>>>> I'm having some trouble getting consistent results with
>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>
>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>> possible.
>>>>>>>
>>>>>>> Can anyone point me in the right direction?
>>>>>>>
>>>>>>>
>>>>>>>     session =
>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>> "/1" .. number_to_call);
>>>>>>>
>>>>>>>     while(session:ready()) do
>>>>>>>        session:answer();
>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>
>>>>>>>        --playbackMessage(check_message);
>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>> get_digits_message, "", "[1]");
>>>>>>>
>>>>>>
>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>
>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>> get_digits_message, "", "\\d+");
>>>>>>
>>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>>> you.
>>>>>> -MC
>>>>>>
>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>
>>>>>>
>>>>>>
>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>
>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>        if(check_success) then
>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>> received from this check.\n");
>>>>>>>           break;
>>>>>>>        else
>>>>>>>           -- not a positive response
>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>           if(attempts == max_attempts) then
>>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>>> attempts for this number.\n");
>>>>>>>             break;
>>>>>>>           end
>>>>>>>        end
>>>>>>>     end
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From delorenzodesign at gmail.com  Tue Mar 16 19:05:01 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Tue, 16 Mar 2010 22:05:01 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
Message-ID: <5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>

So I'm making calls using VoiceNetwork, are you suggesting there's a problem
there?  Or on the line where the call is received (I've tried on a couple of
different networks/providers)?

On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> You inbound DID Provider
>
>
>
> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <
> delorenzodesign at gmail.com> wrote:
>
>> What do you mean about the provider?  I'm not sure I understand that
>> part.  I'm trying to call a number and have the call's recipient type the
>> digits in.
>>
>>
>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> here is the test script I used
>>>
>>> session:answer();
>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter-digits.wav",
>>> "", "\\d+");
>>>
>>> Here is the console log
>>>
>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>> playing file
>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 1:800*
>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 2:640*
>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 3:800*
>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>> 4:640*
>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>>> [1234][\d+]
>>>
>>> You may have some issue with the provider. Check using a local extension.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> Ok, I updated the console log level, and there are no DTMF values being
>>>> detected:
>>>>
>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>> sofia/external/19736327407!
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>> sofia/external/19736327407 entering state [completing][200]
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>> v=0
>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>> s=session
>>>> c=IN IP4 74.51.38.156
>>>> t=0 0
>>>> m=audio 15150 RTP/AVP 0 101
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>> sofia/external/19736327407 entering state [ready][200]
>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>> [sofia/external/19736327407] has been answered
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107 Originate
>>>> Resulted in Success: [sofia/external/19736327407]
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>> CS_SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>> signal sofia/external/19736327407 [BREAK]
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>> Activated L16 at 8000hz 1 channels 20ms
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>> confirmed.
>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>> playing file
>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>> during call from playAndGetDigits:
>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>> out? no
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <
>>>> lloyd.aloysius at gmail.com> wrote:
>>>>
>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>> console log , while you are pressing the keys.
>>>>>
>>>>> Lloyd
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> So I tried Michael's suggestion of:
>>>>>>
>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>> get_digits_message, "", "\\d+");
>>>>>>
>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>> being detected?
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <
>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>
>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>
>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>> possible.
>>>>>>>>
>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>
>>>>>>>>
>>>>>>>>     session =
>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>> "/1" .. number_to_call);
>>>>>>>>
>>>>>>>>     while(session:ready()) do
>>>>>>>>        session:answer();
>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>
>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>
>>>>>>>
>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>
>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>
>>>>>>> Let us know if that works. Oh, one other thing: you might want to add
>>>>>>> an invalid message because playAndGetDigits will handle invalid input for
>>>>>>> you.
>>>>>>> -MC
>>>>>>>
>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>
>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>        if(check_success) then
>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>> received from this check.\n");
>>>>>>>>           break;
>>>>>>>>        else
>>>>>>>>           -- not a positive response
>>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>             freeswitch.consoleLog("info", "We've reached our maximum
>>>>>>>> attempts for this number.\n");
>>>>>>>>             break;
>>>>>>>>           end
>>>>>>>>        end
>>>>>>>>     end
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From pavera at gmail.com  Tue Mar 16 19:17:13 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 20:17:13 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
Message-ID: <82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>

On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>
>> Hello,
>> Pretty new, but so far very happy with freeswitch!  The setup and config
>> is really nice, straightforward, and I appreciate the design of the system.
>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>> really looking for something more stable, and I think I've found it here!
>>
>> One issue I've run into, and I'm not sure if its just that my use case is
>> abnormal, hence the email.
>>
>> I've worked with PBX systems for a number of years (11 now...), and in all
>> of the systems I've worked with, call parking was kind of a "roaming"
>> feature.  By this I mean, you think party X (the intended recipient of the
>> call) is in the building, but they are not at their desk.  So, you park the
>> call, and then page, or by some other means attempt to locate party X and
>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>> party X or you can't find them in time, the call should time out back to the
>> parking party (receptionist) to be handled appropriately (message taken,
>> sent to voicemail, transferred to party X's boss)
>>
>> This scenario requires a few things:
>> 1) the receptionist/person who actually receives the call needs to be able
>> to put the call in a specific place (extension) that doesn't change
>> 2) Party X (the intended recipient) needs to be able to access that
>> specific place (extension) from any phone inside the company
>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>> inadequate, because if the receptionist parks 2 people on 5900, one for
>> party X and one for party Y, in that order, but party Y gets to a phone
>> first, they will be connected to party X's caller)
>> 4) The receptionist has to be protected from or provided with sufficient
>> information to avoid connecting 2 customers with each other (IE the
>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>> what can she do?  What if she just forgets and sends that caller to 6001?
>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>> there...
>>
>> The closest I can get to implementing the above scenario is by using
>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>> receptionist phone to "notify" her that someone is already parked in a
>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>> system, and if she fat fingers her transfer, she will double up a queue,
>> creating the problem indicated in 3 above.  The other shortcoming of this
>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>> becomes impossible to appropriately handle call park.  If the receptionist
>> is away from her desk and attempts to answer a call, she can't tell which
>> parking spots are free at all on a regular 2 line phone.  Also, regular
>> users are prohibited from using call park as they cannot know which slots
>> are free (the need for this arises often in certain industries, such as law,
>> client is speaking with attorney, attorney finishes and says "let me connect
>> you with my assistant to schedule an appointment and handle x, y, and z for
>> you.", paralegal is not at desk, atty needs to park call and page his
>> assistant, maybe they are in the copy room, or the library doing
>> research...)
>>
>> The asterisk parking lot seems to solve this problem quite elegantly
>> (although the stability issues of asterisk undermine it).  It is simple to
>> use (receptionist and everyone else only needs to remember 1 extension to
>> park any call), The callers/parking spots are maintained by the system to
>> prevent double parking, and each caller sits at a specific extension until
>> they are picked up or time out.
>>
>> So, the question boils down to how do people using freeswitch view/use
>> call park?  Is there a "better" roaming feature that I just don't know
>> about?  If it is just something that hasn't been implemented/thought about
>> this way yet, any idea what it would take to get a call parking feature that
>> satisfies these requirements into freeswitch?
>>
>> Thanks for your time, sorry for the long email, hopefully it explains well
>> enough what I'm trying to accomplish.
>>
>> -Tom
>>
>
> Tom,
>
> First of all, welcome to FreeSWITCH - the future of telephony! :P Secondly,
> thank you for actually looking at the documentation before asking your
> question.
>
> You've got yourself quite the little conundrum. Many of us use park in a
> more limited role, or we rely on the receptionist not to do silly things
> like sending one external caller to the slot of another external caller. We
> all agree that's a bad idea, unless it is done deliberately to connect two
> external parties. It sounds to me like this particular situation needs an
> operator panel app where the receptionist can have a pretty web page showing
> what calls are parked where. To my knowledge, this does not yet exist in
> FreeSWITCH. The good news is that FS provides all the requisite building
> blocks to create such an application using the event socket. How do you feel
> about doing a little development? :)
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Michael,
Thanks for the reply. I worked on it more with the business in question
today and we came up with a hybrid solution which I think will cover all the
bases.
1) I implemented a "call park" extension on each phone, these are standard
mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
freeswitch config here BTW, like 8 lines in the dialplan handles this for
all 50 extensions).  This way everyone has a call park extension that can be
picked up anywhere, the numbering scheme is essentially take your extension
prepend a 2, that is your call park extension, so if someone pages you, you
just pick up any phone and dial 2+ext and that will pick the call up.
 Everyone in the office can use this feature to park a call for anyone,
because only calls for a specific person will ever be parked in that queue,
you can put multiple people there without issue.
2) Implemented as mentioned a "parking lot" set of mod_fifo queues that have
SLA/BLA set up on the receptionist phone.  She can use these queues to
quickly and easily park a bunch of people at once (IE when there are call
floods) and then send them off to where they need to go as time permits.  If
she isn't at her phone, she can still answer calls and park them onto
individual users park extensions.  Still doesn't have the "safety" of not
double parking someone in the parking lot, but we'll trust her to be able to
see red lights :)

This whole thing goes live Thursday, so wish me luck :)
Thanks again!
Tom
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From pavera at gmail.com  Tue Mar 16 19:18:46 2010
From: pavera at gmail.com (Tom Christensen)
Date: Tue, 16 Mar 2010 20:18:46 -0600
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
Message-ID: <82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_fifo#Dialplan_Example
At least with mod_fifo (the call parking solution I'm using) there is a
pretty clear example in the wiki.

On Tue, Mar 16, 2010 at 6:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> when a user parked to a slot how to handle the park time out. Is there any
> way to transfer to the call back to a receptionist.
>
> Thanks
> Lloyd
>
>
>
> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 16 20:38:58 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 16 Mar 2010 20:38:58 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
Message-ID: <D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>

Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
-MC

Sent from my iPhone

On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
 > wrote:

> So I'm making calls using VoiceNetwork, are you suggesting there's a  
> problem there?  Or on the line where the call is received (I've  
> tried on a couple of different networks/providers)?
>
> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> You inbound DID Provider
>
>
>
> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> What do you mean about the provider?  I'm not sure I understand that  
> part.  I'm trying to call a number and have the call's recipient  
> type the digits in.
>
>
> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> here is the test script I used
>
> session:answer();
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#", "enter- 
> digits.wav", "", "\\d+");
>
> Here is the console log
>
> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done  
> playing file
> 2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 1:800
> 2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 2:640
> 2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 3:800
> 2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF  
> 4:640
> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test  
> Regex [1234][\d+]
>
> You may have some issue with the provider. Check using a local  
> extension.
>
>
> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> Ok, I updated the console log level, and there are no DTMF values  
> being detected:
>
> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer  
> [soft] 160 bytes per 20ms
> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer  
> sofia/external/19736327407!
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel sofia/ 
> external/19736327407 entering state [completing][200]
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
> v=0
> o=root 21648 21649 IN IP4 74.51.38.156
> s=session
> c=IN IP4 74.51.38.156
> t=0 0
> m=audio 15150 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel sofia/ 
> external/19736327407 entering state [ready][200]
> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel [sofia/ 
> external/19736327407] has been answered
> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107  
> Originate Resulted in Success: [sofia/external/19736327407]
> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497 (sofia/ 
> external/19736327407) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send  
> signal sofia/external/19736327407 [BREAK]
> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec  
> Activated L16 at 8000hz 1 channels 20ms
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354  
> (sofia/external/19736327407) State SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200  
> sofia/external/19736327407 Standard SOFT_EXECUTE
> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354  
> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port  
> confirmed.
> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done  
> playing file
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits  
> received during call from playAndGetDigits:
> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check  
> out? no
>
>
> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> set the console log level to 7. Then you can see the DTMF values in  
> console log , while you are pressing the keys.
>
> Lloyd
>
>
> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> So I tried Michael's suggestion of:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",  
> get_digits_message, "", "\\d+");
>
> but it still didn't detect that I had pressed any keys.  What would  
> cause this?  Is there someplace I can check to see if anything at  
> all is being detected?
>
>
> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <delorenzodesign at gmail.com 
> > wrote:
> I'm having some trouble getting consistent results with  
> playAndGetDigits, sometimes the digits are record and other times  
> they're not.  It seemed to have something to do with how quickly the  
> digits were pressed, but now that no longer seems to be the case.
>
> I'm trying to have users press a confirmation code, in this case  
> "1111" although I'd like to make it accept "1" or "11" or "111" or  
> "1111" if possible.
>
> Can anyone point me in the right direction?
>
>
>     session = freeswitch.Session("{ignore_early_media=true}sofia/ 
> gateway/" .. provider .. "/1" .. number_to_call);
>
>     while(session:ready()) do
>        session:answer();
>        session:setInputCallback("onInput","true");
>
>        --playbackMessage(check_message);
>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",  
> get_digits_message, "", "[1]");
>
> You are specifying a minimum of four and a maximum of four digits.  
> Also, your regex will match any string that has a digit 1 in it,  
> which may or may not be what you are looking for. Lastly, you  
> haven't specified a terminator key (like "#") but you are giving the  
> user only one try and waiting 5000ms for the entry. Try something  
> like this:
>
> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",  
> get_digits_message, "", "\\d+");
>
> Let us know if that works. Oh, one other thing: you might want to  
> add an invalid message because playAndGetDigits will handle invalid  
> input for you.
> -MC
>
> P.S. - This topic is covered in greater detail in Chapter 7 of the  
> upcoming FreeSWITCH book. ;)
>
>
>        freeswitch.consoleLog("info", digits .. "\n");
>        freeswitch.consoleLog("info", "Did this check out? " ..  
> (check_message == true and "yes" or "no" .. "\n"));
>
>        -- we got what we're looking for, so we can end this
>        if(check_success) then
>           freeswitch.consoleLog("info", "A positive response was  
> received from this check.\n");
>           break;
>        else
>           -- not a positive response
>           -- if we reached our max attempts, we're finished move on
>           if(attempts == max_attempts) then
>             freeswitch.consoleLog("info", "We've reached our maximum  
> attempts for this number.\n");
>             break;
>           end
>        end
>     end
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
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From infos at madovsky.org  Tue Mar 16 21:20:15 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 00:20:15 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
Message-ID: <3E1709744536417C877AB5952395DAED@MOBILEE1705>

Also Reddy,

FS is protected (correct me if I'm wrong)
under MPL licence
http://www.mozilla.org/MPL/MPL-1.1.html

in the article 3, paragraph 2:
3.2. Availability of Source Code.
Any Modification which You create or to which You contribute must be made available in Source Code form under the terms of this License either on the same media as an Executable version or via an accepted Electronic Distribution Mechanism to anyone to whom you made an Executable version available; and if made available via Electronic Distribution Mechanism, must remain available for at least twelve (12) months after the date it initially became available, or at least six (6) months after a subsequent version of that particular Modification has been made available to such recipients. You are responsible for ensuring that the Source Code version remains available even if the Electronic Distribution Mechanism is maintained by a third party. 

so my opinion is you are infringing the Licence 

Regards

Franck

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 12:00 PM
  Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


  so share your customized version to the community and maybe
  we ll be glad to help you.... this is the principle of open source
    ----- Original Message ----- 
    From: srinivasula reddy 
    To: freeswitch-users 
    Sent: Tuesday, March 16, 2010 11:49 AM
    Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


    we started customizing Freeswitch1.0.2 2yrs back, so now we are unable to use new versions.

    srinivas


    On Mon, Mar 15, 2010 at 9:06 PM, srinivasula reddy <srinivas.ksvreddy at gmail.com> wrote:

      Hi,

      i am using freeswitch1.0.2, farwarding call from freeswitch to cisco like just farwading the call. 
      <action application="bridge" data=sofia/internal/${destination_number}@ciscogatewayip/>
      i do not using any gateway here, just farwarding call to router, is this correct way of doing? any idea

      -- 
      Srinivasula Reddy K




    -- 
    Srinivasula Reddy K



----------------------------------------------------------------------------


    _______________________________________________
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From jaybinks at gmail.com  Tue Mar 16 21:30:26 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 17 Mar 2010 14:30:26 +1000
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
Message-ID: <ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>

Im not on the FS dev team, but I think you may be a little mistaken.

generally your free to make modifications to open source code ( for
internal use )
however its when you distribute the resultant product, then you have
to make the source available..

if your only using it in-house, I dont believe your required to
contribute the code back.
( however your going to end up having issues maintaining your patch
set over a long period.. like can be seen here )

see here :

> must be made available in Source Code form under the terms of this License either on the
> same media as an Executable version

if your not distributing your executable to any 3rd parties... then I
dont believe you have to distribute the modified source..

Correct me if Im wrong though guys...

Jay



From maciej.aniserowicz at gmail.com  Wed Mar 17 00:39:40 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 16 Mar 2010 23:39:40 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
Message-ID: <1268811580808-4748784.post@n2.nabble.com>


Hi again,
I don't quite understand why both of them would not be parked... in logs
(from http://pastebin.freeswitch.org/12432) I can see this:

dialing 1st call
[5] 2010-03-15 08:52:45.191854 [NOTICE] switch_channel.c:666 New Channel
sofia/external/2500 [b278ec65-6d83-4454-b885-7ee66f414230]

then...:
[55] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[56] EXECUTE sofia/external/2500 park()

then...:
[57] 2010-03-15 08:52:45.332056 [DEBUG] switch_ivr.c:1426
(sofia/external/2500) State Change CS_EXECUTE -> CS_ROUTING
[58] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_session.c:1018 Send
signal sofia/external/2500 [BREAK]
[59] 2010-03-15 08:52:45.332056 [DEBUG] switch_core_session.c:638 Send
signal sofia/external/2500 [BREAK]
[60] 2010-03-15 08:52:45.332056 [NOTICE] switch_ivr.c:1432 Transfer
sofia/external/2500 to inline[park at default]

then PARK again...:
[72] 2010-03-15 08:52:45.342070 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[73] EXECUTE sofia/external/2500 park()
[74] 2010-03-15 08:52:45.402157 [DEBUG] switch_rtp.c:2055 Correct ip/port
confirmed.
[75] 2010-03-15 08:52:50.709789 [DEBUG] switch_ivr.c:1426
(sofia/external/2500) State Change CS_EXECUTE -> CS_ROUTING

then TRANSFER again...:
[78] 2010-03-15 08:52:50.709789 [NOTICE] switch_ivr.c:1432 Transfer
sofia/external/2500 to inline[park at default]

and PARK again...:
[90] 2010-03-15 08:52:50.719803 [DEBUG] switch_core_state_machine.c:157
sofia/external/2500 Standard EXECUTE
[91] EXECUTE sofia/external/2500 park()


And the same goes for the 2nd channel. The Transfer/Park instructions are
repeated several times. Is this normal? My command is fine according to WIKI
I guess:

bgapi originate
{originate_timeout=15}[24cc_session_id=76dfb907-a0c3-4357-b1f9-ede89caf8a7e,ignore_early_media=true,max_forwards=999999,origination_uuid=d4f4e460-43c7-4e53-af18-46fa5226227d]sofia/gateway/agent_gateway/2500
&park()

Or maybe the &park() is not necessary there?

This mechanism worked mostly OK for several months with previous builds of
freeswitch, the problems showed up after upgrading to trunk recently.
Sometimes it works though, i'd say that ~30% of the calls end up being
hanged up, but most of them are just fine.

Anyway, I'll try to wait for CHANNEL_PARK event as you suggested and only
then assume that the call is really "answered" and ready for processing.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4748784.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From robert.dyck at shaw.ca  Wed Mar 17 00:49:05 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 00:49:05 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
Message-ID: <201003170049.06565.robert.dyck@shaw.ca>

Is there something wrong with my build?

fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting [Socket 
Connection Error]" with fscomm running.

RD

On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> With FSComm running you can use fs_cli to try pa list. If u can't list the
> devices that would be the problem. I'm going to try to dig into this
> further on 1 of my desktops w/ pulseaudio.
> 
> -- W



From brian at microcomaustralia.com.au  Wed Mar 17 01:17:18 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 17 Mar 2010 19:17:18 +1100
Subject: [Freeswitch-users] dahdi problems
Message-ID: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>

Hello,

Just noticed zaptel was renamed dahdi.

What am I doing wrong?

2010-03-17 19:11:32.505233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/modules.conf.
2010-03-17 19:11:32.506233 [NOTICE] zap_io.c:3048 Modules configured: 1
2010-03-17 19:11:32.506233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/openzap.conf.
2010-03-17 19:11:32.506233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.506233 [NOTICE] ozmod_zt.c:1182 Using DAHDI control device
2010-03-17 19:11:32.507233 [INFO] zap_io.c:2849 Loading IO from
/opt/freeswitch/mod/ozmod_zt.so [zt]
2010-03-17 19:11:32.507233 [DEBUG] zap_config.c:56 Configuration file
is /opt/freeswitch/conf/zt.conf.
2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000
2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000
2010-03-17 19:11:32.508233 [INFO] zap_io.c:2595 auto-loaded 'zt'
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2616 created span 1 (FXS1)
of type zt
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [name]=[Brian May]
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [number]=[1002]
2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [fxs-channel]=[1]
2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2671 setting trunk type to
'FXS' start(KEWL)
2010-03-17 19:11:32.509233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 1 fd 64 (No such device or address)
2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2616 created span 2 (FXS2)
of type zt
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [name]=[Brian May]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [number]=[1002]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [fxs-channel]=[2]
2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2671 setting trunk type to
'FXS' start(KEWL)
2010-03-17 19:11:32.511233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 2 fd 64 (No such device or address)
2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2616 created span 3 (FXO1)
of type zt
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [name]=[OpenZAP]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [number]=[0397551926]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [fxo-channel]=[3]
2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2658 setting trunk type to
'FXO' start(KEWL)
2010-03-17 19:11:32.512233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 3 fd 64 (No such device or address)
2010-03-17 19:11:32.512233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.513233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2578 found config for span
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2616 created span 4 (FXO2)
of type zt
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [name]=[OpenZAP]
2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [number]=[0397552345]
2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2629 span 4 [fxo-channel]=[4]
2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2658 setting trunk type to
'FXO' start(KEWL)
2010-03-17 19:11:32.514233 [ERR] ozmod_zt.c:269 failure configuring
device /dev/dahdi/channel chan 4 fd 64 (No such device or address)
2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not
support command 32!
2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not
support command 34!
2010-03-17 19:11:32.515233 [INFO] zap_io.c:2773 Configured 0 channel(s)
2010-03-17 19:11:32.516233 [ERR] zap_io.c:3056 No modules configured!
2010-03-17 19:11:32.516233 [ERR] mod_openzap.c:3171 Error loading OpenZAP
2010-03-17 19:11:32.516233 [CRIT] switch_loadable_module.c:882 Error
Loading module /opt/freeswitch/mod/mod_openzap.so
**Module load routine returned an error**


Is there anything else I need to change when going from zaptel to dahdi?

The kernel modules seem to load ok, and /dev/dahdi/* looks good too.
-- 
Brian May <brian at microcomaustralia.com.au>



From irmatov at gmail.com  Wed Mar 17 01:32:50 2010
From: irmatov at gmail.com (Timur Irmatov)
Date: Wed, 17 Mar 2010 13:32:50 +0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
Message-ID: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>

On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org> wrote:
> It already works use the respond application in the dialplan to respond 484

Thanks for the hint, Brian. I have a SoftX3000 softswitch sending
calls to FreeSWITCH. SoftX3000 is configured to send anything between
5 and 20 digits. I want FreeSWITCH to respond 484 until it receives
enough digits to make a call.

Here is an excerpt from my dialplan:

<extension name="public_extensions">
  <condition field="destination_number" expression="^006[0-9]{0,13}$">
    <action application="respond" data="484"/>
  </condition>
</extension>

In the first INVITE, softswitch sends 5 digits: 00610. FreeSWITCH
answers 484 just as I want it to. Softswitch sends another INVITE,
this time with 6 digits: 006107, but *without* SDP. Missing SDP upsets
FreeSWITCH and it responds 480, call finishes. Here is log from
FreeSWITCH:

2010-03-17 09:45:07.329047 [NOTICE] switch_channel.c:669 New Channel
sofia/external/1220139 at 10.0.2.5 [dc6f5d24-317f-11df-9775-c1f542b963a5]
2010-03-17 09:45:07.331047 [INFO] mod_dialplan_xml.c:418 Processing
1220139->00610 in context public
2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:185
sofia/external/1220139 at 10.0.2.5 has executed the last dialplan
instruction, hanging up.
2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:187
Hangup sofia/external/1220139 at 10.0.2.5 [CS_EXECUTE] [NORMAL_CLEARING]
2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1179 Session
3 (sofia/external/1220139 at 10.0.2.5) Ended
2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
2010-03-17 09:45:08.448839 [NOTICE] switch_channel.c:669 New Channel
sofia/external/1220139 at 10.0.2.5 [dd1a3776-317f-11df-9776-c1f542b963a5]
2010-03-17 09:45:08.448839 [INFO] sofia.c:4388 No SDP in INVITE and
3pcc not enabled, hanging up.
2010-03-17 09:45:08.449840 [NOTICE] sofia.c:4390 Hangup
sofia/external/1220139 at 10.0.2.5 [CS_NEW] [MANDATORY_IE_MISSING]
2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1179 Session
4 (sofia/external/1220139 at 10.0.2.5) Ended
2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]


And here are SIP messages, capture made with ngrep:

U 10.0.2.5:5070 -> 172.16.12.11:5090
INVITE sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
Supported: 100rel.
P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
Content-Length: 263.
Content-Type: application/sdp.
.
v=0.
o=HuaweiSoftX3000 98044 98044 IN IP4 10.0.2.5.
s=Sip Call.
c=IN IP4 172.16.12.1.
t=0 0.
m=audio 50000 RTP/AVP 8 0 18 4 2.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:2 G726-32/8000.
a=fmtp:18 annexb=yes.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
CSeq: 1 INVITE.
Contact: <sip:00610 at 172.16.12.11:5090;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.	
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Length: 0.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
ACK sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
CSeq: 1 ACK.
Max-Forwards: 70.
Content-Length: 0.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
INVITE sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>.
CSeq: 1 INVITE.
Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Content-Length: 0.
.

#
U 172.16.12.11:5090 -> 10.0.2.5:5070
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
CSeq: 1 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING".
Content-Length: 0.
P-Asserted-Identity: "006107" <sip:006107 at 172.16.12.11>.
.

#
U 10.0.2.5:5070 -> 172.16.12.11:5090
ACK sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
CSeq: 1 ACK.
Max-Forwards: 70.
Content-Length: 0.

RFC3578 says:

"The new INVITE has the same Call-ID and the same From header field
including the tag as the first INVITE sent, but has an updated
Request-URI."

SoftX3000 sends second INVITE with different Call-ID. Is there
anything I can do on FreeSWITCH to solve this problem and continue
receiving additional digits?


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From dujinfang at gmail.com  Wed Mar 17 03:01:43 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 17 Mar 2010 18:01:43 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
Message-ID: <23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>

2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> Are you bridging to sip or some ivr and making dingaling wait until the call
> answers?
>

no I was originate dingaling/... &echo,   also tried originate
dingaling/...  extension XML public (which will hit dialplan and
answer() after client answers), neither work.


> gtalk is sort of missing key telephony features like the concept of pdd and
> indication progress etc.
>
> Try answering the call in your dialplan before you bridge.
>
> I think the gtalk client is getting impatient because you did not answer
> soon enough with an ip choice
> so it starts sending all these extra ones at you and gave up on the one we
> chose.
>
>

I tested on another server runing FreeSWITCH Version 1.0.trunk
(16958M), it works.

see http://pastebin.freeswitch.org/12463

interesting, messages like the same, but I see the working version has
one more line on line 174:

http://pastebin.freeswitch.org/12463 line: 173 to 175
2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
[dingaling/gmail.com/seven at idapted.com] has been answered
2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
(dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
CS_ROUTING


http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)

2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
(dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
CS_ROUTING

Do you see any other difference? Thanks.


> On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > FS would prefer you run your kernel at 1000hz you are probably on a VM
>> > if
>> > its 10000hz
>> >
>>
>> It's a ?physical machine. And I'm also running 20 skype clients. as
>> 100HZ is recommended for skypiax. I think the kernel should be 100 or
>> 250 HZ. weird.
>>
>>
>> here is the dingaling log with dl_debug on
>>
>> http://pastebin.freeswitch.org/12453
>> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>>
>>
>> still get this message on client:
>>
>> Sorry! The voice chat with language_lesson failed because of a network
>> problem at 5:39 PM. Please try again.
>> ?Click here to report this to Google
>>
>>
>>
>> Thanks.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> >
>> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> >> Windows client and web site on Mac. They both work on the other two
>> >> instances of FreeSWITCH I mentioned in this thread.
>> >>
>> >> btw, can you give me a hint on the timer warning?
>> >>
>> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> > you would have to execute
>> >> >
>> >> > dl_debug on
>> >> >
>> >> > to get a trace that would tell us anything about that?
>> >> >
>> >> > Are you using the windows gtalk client or the web site?
>> >> > We are following google who have been pushing for the online client
>> >> > embedded
>> >> > in the gmail website.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> here is the client:
>> >> >>
>> >> >> <profile type="client">
>> >> >> ?<param name="name" value="gmail.com"/>
>> >> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> >> ?<param name="password" value="password"/>
>> >> >> ?<param name="server" value="talk.google.com"/>
>> >> >> ?<param name="dialplan" value="XML"/>
>> >> >> ?<param name="contex" value="public"/>
>> >> >> ?<param name="message" value="FreeSWITCH"/>
>> >> >> ?<param name="rtp-ip" value="auto"/>
>> >> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> >> ?<param name="auto-login" value="true"/>
>> >> >> ?<param name="auto-reply" value="Welcome"/>
>> >> >> ?<param name="sasl" value="plain"/>
>> >> >> ?<param name="tls" value="true"/>
>> >> >> ?<param name="exten" value="5000"/>
>> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> >> </profile>
>> >> >>
>> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> >> > ALL,
>> >> >> >
>> >> >> > I just upgrade FS to 16992.
>> >> >> >
>> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
>> >> >> > sounds
>> >> >> > ok.
>> >> >> >
>> >> >> >
>> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000 microseconds
>> >> >> > detected!
>> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> >> > experience audio problems.
>> >> >> >
>> >> >> > uname -a
>> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12 08:27:05
>> >> >> > UTC
>> >> >> > 2008 i686 GNU/Linux
>> >> >> >
>> >> >> >
>> >> >> > I used a script to check the interupts
>> >> >> >
>> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >> >
>> >> >> >
>> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
>> >> >> > answer.
>> >> >> > I
>> >> >> > originate a call from FS, the console blocks, gtalk client ring
>> >> >> > and
>> >> >> > it
>> >> >> > looks ok when I hit answer, but no sound, FS console still blocks.
>> >> >> > a
>> >> >> > few seconds later gtalk client shows:
>> >> >> >
>> >> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> >> > network
>> >> >> > problem at 5:50 PM. Please try again.
>> >> >> > ?Click here to report this to Google.
>> >> >> >
>> >> >> >
>> >> >> > here is a log with bgapi:
>> >> >> > http://pastebin.freeswitch.org/12434
>> >> >> >
>> >> >> >
>> >> >> > rev 14696 works on the same server same conf. another server
>> >> >> > running
>> >> >> > 16958M works at the same time. I have no clue to check, is it
>> >> >> > related
>> >> >> > to the timer as in 1)? can some help me to take a look? let me
>> >> >> > know
>> >> >> > if
>> >> >> > you need "dl_debug on" logs.
>> >> >> >
>> >> >> > Thanks.
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From ivdreg at gmail.com  Wed Mar 17 03:42:20 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 12:42:20 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
Message-ID: <b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>

Thanks again Rupa,

Is it possible to have more than one limit per call ? I have an issue in
this case with calls per period limiting.

This is part of my dial-plan before bridge:

<action application="limit_hash" data="caller ${destination_number} 5
!INCOMING_CALL_BARRED" />
<action application="limit_hash" data="crcd_${destination_number}
${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
<action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />

These are vars for 3 calls (not more than 10 sec between them):

variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]
--
variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]
--
variable_limit_usage_caller_111111111111: [1]
variable_limit_rate_caller_111111111111: [0]
variable_limit_usage_crcd_111111111111_99999999999: [1]
variable_limit_rate_crcd_111111111111_99999999999: [0]
variable_limit_usage: [1]
variable_limit_usage_cr_TEST_99999999999: [1]
variable_limit_rate: [1]
variable_limit_rate_cr_TEST_99999999999: [1]

I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
case. Is it true ? Does limit_usage and limit_rate that have no realm need
to exist also ?

Thanks

2010/3/16 Rupa Schomaker <rupa at rupa.com>

> limit and limit_hash should work the same with regards to disconnect cause.
>  If you see a difference, let me know.
>
>
> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Thanks Rupa,
>>
>> This was good clarification. I want to use mod_limit via 4 servers but
>> because traffic is equally balanced between them I will use limit_hash and
>> calculate how many calls every server will serve.
>>
>> One more question ? Is it necessary limit to transfer to limit_exceeded or
>> there is a way to set directly disconnect cause like in limit_hash ?
>>
>> Cheers
>>
>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>
>> 1) limit_hash use an in-memory hash for it's storage. There is nor storing
>>> to a database when using limit_hash.  If you want to use the database, then
>>> use plain limit.
>>>
>>> 2) If you want a max calls in a 24hrs period use you rate with
>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>
>>> limit_hash realm id 50/86400
>>>
>>> should do what you want -- from memory.  double check the wiki/help to
>>> verify syntax.
>>>
>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>
>>>> Hi All,
>>>>
>>>> I have some questions regarding mod_limit.
>>>>
>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>
>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>
>>>> and they are applied on inbound call profile in same extension.
>>>>
>>>>
>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>> call but not calls processed as count. Any idea ?
>>>>
>>>> Thanks for any help
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Wed Mar 17 03:57:52 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 17 Mar 2010 05:57:52 -0500
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
References: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
Message-ID: <4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>

Brian May wrote:

> Is there anything else I need to change when going from zaptel to dahdi?
> 
> The kernel modules seem to load ok, and /dev/dahdi/* looks good too.

Make the appropriate changes to the configuration files in /etc/dahdi/. Take a look at the files in /proc/dahdi/.

--
Russell Mosemann




From ivdreg at gmail.com  Wed Mar 17 04:03:42 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 13:03:42 +0200
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
References: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
Message-ID: <b437a0e1003170403yc389c99q433156994fb51c46@mail.gmail.com>

Hi this is a little example from my perl code that do that:

array random_be contains randomized IP addresses of next hops.
there also hash:
my %hunt_error=(
        NO_USER_RESPONSE => 1,
        NETWORK_OUT_OF_ORDER => 1,
        DESTINATION_OUT_OF_ORDER => 1,
        SYSTEM_SHUTDOWN => 1,
);

If error is defined in this array we hunt else drop call. I assume that
these errors are internally generated from this FS instance or comes first
remote hop. So I need to hunt over next hop. But if error comes from last
server I drop call and do not hunt over rest of my SIP servers.

for (my $i=0;$i <= scalar(@random_be) ; $i++) {
        my $dst_gw = shift(@random_be);
        print "GW:[$dst_gw]\n";
        $bridge = "sofia/internal_out/$called\@$dst_gw";
        $session->execute('bridge',"$bridge");
        $berror = $session->getVariable("originate_disposition");
        if (defined($berror)){
                fprint("\tBridge Error - Check continue or not\n");
                if (!defined($hunt_error{$berror})){
                        fprint("\t HANGUP - ERROR NOT DEFINED");

$session->execute('export','continue_on_fail=false');
                        $session->hangup();
                        return (1);
                }
        } else { last; } ;  ### NO originate_disposition so successful
bridge
}

Suppose this will help. Not in Lua but idea is the same.

Cheers


2010/3/14 rentmycoder rentmycoder <rentmycoder at gmail.com>

> > Pass them as arguments to your luarun hangup hook.
> And what about variables are not available before the call setup (like
> uuid), only during the call progress or after the hangup???
> I would need to access the originate_disposition variable...
>
> >> 'luarun hangup.lua ${originate_disposition}'
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> any other hint???
>
>
> > ---------- Forwarded message ----------
> > From: Brian West <brian at freeswitch.org>
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Sun, 14 Mar 2010 13:00:10 -0500
> > Subject: Re: [Freeswitch-users] how to access session variables after
> hangup???
> > Pass them as arguments to your luarun hangup hook.
> >
> > /b
> >
> > On Mar 14, 2010, at 12:40 PM, rentmycoder rentmycoder wrote:
> >
> >> Hi guys,
> >> Freeswitch rocks!
> >> except the documentation:(
> >> I'm not able to figure out how to access session variables after
> hangup...
> >> I try to dial out on a gateway and detect it the called party hangs up
> >> before bridge...
> >>
> >> bgapi originate
> >>
> {ignore_early_media=true,continue_on_fail=true,bypass_media=false,hangup_after_bridge=true,originate_timeout=10,api_hangup_hook='luarun
> >> hangup.lua ${uuid}'}sofia/gateway/phoneno &park()
> >>
> >> I cannot access the variables using:
> >> 1. script arguments:
> >> 'luarun hangup.lua ${uuid}'}
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> >> (${api_hangup_hook} contains a variable)
> >>
> >> 2. session:getVariable("originate_disposition")
> >> 2010-03-14 18:31:37.042756 [ERR] mod_lua.cpp:182 hangup.lua:2: attempt
> >> to call global 'getVar' (a nil value)
> >>
> >> 3. params:getHeader("variable_sip_req_uri")
> >> 2010-03-14 18:36:19.973939 [ERR] mod_lua.cpp:182 hangup.lua:3: attempt
> >> to index global 'params' (a nil value)
> >>
> >> I have asked this before on the dev list too, but Anthony didn't gave
> >> me a clean answer here:
> >>
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-July/002491.html
> >>
> >> "In the script you can access the data from the env event which is
> >> automatically created for you and it
> >> contains all the channel variables from that departed channel."
> >>
> >> Ok, but how?????
> >>
> >> Please help...
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Wed Mar 17 05:01:15 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 09:01:15 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003170049.06565.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003161411.43845.robert.dyck@shaw.ca>
	<6b65470d1003161436m1e7d4914r811fca9d70ffb73d@mail.gmail.com>
	<201003170049.06565.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>

Rob, I believe that the core has failed to load. I have not added any type
of check on the code level for those cases yet as I believed it would never
really fail to load. Those cases have been more frequent than what I
originally expected so I will add that to code today (hopefully later
tonight).

Meanwhile, what you can do is check if the ports are open or not on the OS.
I am not sure how to do that on Windows. Or, maybe we already have logs on
the log folder?

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> Is there something wrong with my build?
>
> fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> [Socket
> Connection Error]" with fscomm running.
>
> RD
>
> On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > With FSComm running you can use fs_cli to try pa list. If u can't list
> the
> > devices that would be the problem. I'm going to try to dig into this
> > further on 1 of my desktops w/ pulseaudio.
> >
> > -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xengelpublicx at gmail.com  Wed Mar 17 05:22:11 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Wed, 17 Mar 2010 15:22:11 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
Message-ID: <eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>

Yes

On Wed, Mar 17, 2010 at 12:19 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Mar 15, 2010 at 1:21 PM, Vladimir Elizarov <xengelpublicx at gmail.com>
> wrote:
>>
>> Hello.
>>
>> How can I intercept a fifo? I tried to use the example of intecept
>> default dialplan, but in no fifo dialed_extension.
>
> Are you trying to intercept the call that is ringing at a FIFO agent's
> phone?
> -MC
>
>>
>> ? ?<extension name="operators">
>> ? ? ?<condition field="destination_number" expression="^operators$">
>> ? ? ? ? ? ?<action application="bind_meta_app" data="2 b s
>>
>> record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>> ? ? ? ? ? ?<action application="hash"
>>
>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>> ? ? ? ? ? ?<action application="hash"
>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>> ? ? ? ? ? ?<action application="set"
>>
>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)}"/>
>> ? ? ? ? ? ?<action application="hash"
>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>> ? ? ? ? ? ?<action application="answer"/>
>> ? ? ? ? ? ?<action application="playback"
>> data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/>
>> ? ? ? ? ? ?<action application="playback"
>> data="ivr/ivr-hold_connect_call.wav"/>
>> ? ? ? ? ? ?<action application="set" data="fifo_music=$${hold_music}"/>
>> ? ? ? ? ? ?<action application="fifo" data="operators@$${domain} in
>> undef"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> ? ?<fifo name="operators@$${domain}" importance="0">
>> ? ? ?<member timeout="15" simo="1"
>>
>> lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member>
>> ? ?</fifo>
>> ?</fifos>
>> --
>> Best regards, Vladimir Elizarov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov



From brian at freeswitch.org  Wed Mar 17 06:18:54 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:18:54 -0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
Message-ID: <A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>


On Mar 17, 2010, at 3:32 AM, Timur Irmatov wrote:

> On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org> wrote:
>> It already works use the respond application in the dialplan to respond 484
> 
> 
> RFC3578 says:
> 
> "The new INVITE has the same Call-ID and the same From header field
> including the tag as the first INVITE sent, but has an updated
> Request-URI."
> 
> SoftX3000 sends second INVITE with different Call-ID. Is there
> anything I can do on FreeSWITCH to solve this problem and continue
> receiving additional digits?

NO, Why should we even try?  You paid Huawei for this switch that is doing something wrong.  Contact them and make them fix it... if they won't then you have to ask yourself why they should?  You paid them... they have your money... they don't care.

/b




From brian at freeswitch.org  Wed Mar 17 06:19:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:19:15 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268811580808-4748784.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
Message-ID: <E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>

First off are you on SVN Trunk?

/b

On Mar 17, 2010, at 2:39 AM, Maciej Aniserowicz wrote:

> Hi again,
> I don't quite understand why both of them would not be parked... in logs
> (from http://pastebin.freeswitch.org/12432) I can see this:

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From brian at freeswitch.org  Wed Mar 17 06:23:03 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 08:23:03 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
Message-ID: <EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>

This is where it gets hairy... Yes the MPL says you have to share your modifications with the original author but as far as I can tell you don't have to distribute to cause this clause to go into effect.  

Its like this:

You change switch_ivr_originate.c, you HAVE to share those changes with Anthony.
You create your own mod_kewlshit.c, its yours to do with what you want and you don't have to share.

You change any code that is included with the download YOU MUST share with the original author.

/b


On Mar 16, 2010, at 11:30 PM, jay binks wrote:

> Im not on the FS dev team, but I think you may be a little mistaken.
> 
> generally your free to make modifications to open source code ( for
> internal use )
> however its when you distribute the resultant product, then you have
> to make the source available..
> 
> if your only using it in-house, I dont believe your required to
> contribute the code back.
> ( however your going to end up having issues maintaining your patch
> set over a long period.. like can be seen here )
> 
> see here :
> 
>> must be made available in Source Code form under the terms of this License either on the
>> same media as an Executable version
> 
> if your not distributing your executable to any 3rd parties... then I
> dont believe you have to distribute the modified source..
> 
> Correct me if Im wrong though guys...
> 
> Jay
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rupa at rupa.com  Wed Mar 17 06:41:08 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 17 Mar 2010 08:41:08 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
Message-ID: <b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>

That should work.  Can you open a jira against mod_limit and assign it to
me?  Thanks.

On Wed, Mar 17, 2010 at 5:42 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Thanks again Rupa,
>
> Is it possible to have more than one limit per call ? I have an issue in
> this case with calls per period limiting.
>
> This is part of my dial-plan before bridge:
>
> <action application="limit_hash" data="caller ${destination_number} 5
> !INCOMING_CALL_BARRED" />
> <action application="limit_hash" data="crcd_${destination_number}
> ${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
> <action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
> ${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />
>
> These are vars for 3 calls (not more than 10 sec between them):
>
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
> --
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
> --
> variable_limit_usage_caller_111111111111: [1]
> variable_limit_rate_caller_111111111111: [0]
> variable_limit_usage_crcd_111111111111_99999999999: [1]
> variable_limit_rate_crcd_111111111111_99999999999: [0]
> variable_limit_usage: [1]
> variable_limit_usage_cr_TEST_99999999999: [1]
> variable_limit_rate: [1]
> variable_limit_rate_cr_TEST_99999999999: [1]
>
> I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
> case. Is it true ? Does limit_usage and limit_rate that have no realm need
> to exist also ?
>
> Thanks
>
>
> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>
>> limit and limit_hash should work the same with regards to disconnect
>> cause.  If you see a difference, let me know.
>>
>>
>> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>
>>> Thanks Rupa,
>>>
>>> This was good clarification. I want to use mod_limit via 4 servers but
>>> because traffic is equally balanced between them I will use limit_hash and
>>> calculate how many calls every server will serve.
>>>
>>> One more question ? Is it necessary limit to transfer to limit_exceeded
>>> or there is a way to set directly disconnect cause like in limit_hash ?
>>>
>>> Cheers
>>>
>>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>>
>>> 1) limit_hash use an in-memory hash for it's storage. There is nor
>>>> storing to a database when using limit_hash.  If you want to use the
>>>> database, then use plain limit.
>>>>
>>>> 2) If you want a max calls in a 24hrs period use you rate with
>>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>>
>>>> limit_hash realm id 50/86400
>>>>
>>>> should do what you want -- from memory.  double check the wiki/help to
>>>> verify syntax.
>>>>
>>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>>
>>>>> Hi All,
>>>>>
>>>>> I have some questions regarding mod_limit.
>>>>>
>>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>>
>>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>>
>>>>> and they are applied on inbound call profile in same extension.
>>>>>
>>>>>
>>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>>> call but not calls processed as count. Any idea ?
>>>>>
>>>>> Thanks for any help
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From delorenzodesign at gmail.com  Wed Mar 17 07:00:56 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 17 Mar 2010 10:00:56 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<87f2f3b91003161400j2afaed55sc74a42aa49926bd5@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
Message-ID: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>

How do I do that in Lua?  session:execute("dtmf","start") doesn't work.

On Tue, Mar 16, 2010 at 11:38 PM, Michael S Collins <msc at freeswitch.org>wrote:

> Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
> -MC
>
> Sent from my iPhone
>
> On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <delorenzodesign at gmail.com>
> wrote:
>
> So I'm making calls using VoiceNetwork, are you suggesting there's a
> problem there?  Or on the line where the call is received (I've tried on a
> couple of different networks/providers)?
>
> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
> lloyd.aloysius at gmail.com> wrote:
>
>> You inbound DID Provider
>>
>>
>>
>> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>> delorenzodesign at gmail.com> wrote:
>>
>>> What do you mean about the provider?  I'm not sure I understand that
>>> part.  I'm trying to call a number and have the call's recipient type the
>>> digits in.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> here is the test script I used
>>>>
>>>> session:answer();
>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>> "enter-digits.wav", "", "\\d+");
>>>>
>>>> Here is the console log
>>>>
>>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>>> playing file
>>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 1:800*
>>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 2:640*
>>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 3:800*
>>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>> 4:640*
>>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test Regex
>>>> [1234][\d+]
>>>>
>>>> You may have some issue with the provider. Check using a local
>>>> extension.
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>> delorenzodesign at gmail.com> wrote:
>>>>
>>>>> Ok, I updated the console log level, and there are no DTMF values being
>>>>> detected:
>>>>>
>>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>>> [soft] 160 bytes per 20ms
>>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>>> sofia/external/19736327407!
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>> sofia/external/19736327407 entering state [completing][200]
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>>> v=0
>>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>>> s=session
>>>>> c=IN IP4 74.51.38.156
>>>>> t=0 0
>>>>> m=audio 15150 RTP/AVP 0 101
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=silenceSupp:off - - - -
>>>>> a=ptime:20
>>>>>
>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>> sofia/external/19736327407 entering state [ready][200]
>>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>>> [sofia/external/19736327407] has been answered
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107
>>>>> Originate Resulted in Success: [sofia/external/19736327407]
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>>> CS_SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>>> signal sofia/external/19736327407 [BREAK]
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>>> Activated L16 at 8000hz 1 channels 20ms
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>>> confirmed.
>>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>>> playing file
>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>>> during call from playAndGetDigits:
>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>>> out? no
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>>> lloyd.aloysius at gmail.com> wrote:
>>>>>
>>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>>> console log , while you are pressing the keys.
>>>>>>
>>>>>> Lloyd
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>
>>>>>>> So I tried Michael's suggestion of:
>>>>>>>
>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>
>>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>>> being detected?
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <<msc at freeswitch.org>
>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>>
>>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>>> possible.
>>>>>>>>>
>>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>     session =
>>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>>> "/1" .. number_to_call);
>>>>>>>>>
>>>>>>>>>     while(session:ready()) do
>>>>>>>>>        session:answer();
>>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>>
>>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>>
>>>>>>>>
>>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>>
>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>
>>>>>>>> Let us know if that works. Oh, one other thing: you might want to
>>>>>>>> add an invalid message because playAndGetDigits will handle invalid input
>>>>>>>> for you.
>>>>>>>> -MC
>>>>>>>>
>>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>>
>>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>>        if(check_success) then
>>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>>> received from this check.\n");
>>>>>>>>>           break;
>>>>>>>>>        else
>>>>>>>>>           -- not a positive response
>>>>>>>>>           -- if we reached our max attempts, we're finished move on
>>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>>             freeswitch.consoleLog("info", "We've reached our
>>>>>>>>> maximum attempts for this number.\n");
>>>>>>>>>             break;
>>>>>>>>>           end
>>>>>>>>>        end
>>>>>>>>>     end
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Mar 17 07:14:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 09:14:56 -0500
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
	<A4D63B61-6CE1-4E7A-BD87-AD33CB1CDAB6@freeswitch.org>
Message-ID: <191c3a031003170714s24a14ce0oa58ec91bc0ec84a@mail.gmail.com>

Not to mention we would have to hack the sofia sip stack in ways I can't
begin to imagine.
It so horribly violates the spec its a lost cause.


On Wed, Mar 17, 2010 at 8:18 AM, Brian West <brian at freeswitch.org> wrote:

>
> On Mar 17, 2010, at 3:32 AM, Timur Irmatov wrote:
>
> > On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org>
> wrote:
> >> It already works use the respond application in the dialplan to respond
> 484
> >
> >
> > RFC3578 says:
> >
> > "The new INVITE has the same Call-ID and the same From header field
> > including the tag as the first INVITE sent, but has an updated
> > Request-URI."
> >
> > SoftX3000 sends second INVITE with different Call-ID. Is there
> > anything I can do on FreeSWITCH to solve this problem and continue
> > receiving additional digits?
>
> NO, Why should we even try?  You paid Huawei for this switch that is doing
> something wrong.  Contact them and make them fix it... if they won't then
> you have to ask yourself why they should?  You paid them... they have your
> money... they don't care.
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 07:16:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 09:16:44 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
Message-ID: <191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>

Were you using gtalk and FS both behind nat with the public gtalk server by
any chance?
This setup requires special configuration.


On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:

> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> > Are you bridging to sip or some ivr and making dingaling wait until the
> call
> > answers?
> >
>
> no I was originate dingaling/... &echo,   also tried originate
> dingaling/...  extension XML public (which will hit dialplan and
> answer() after client answers), neither work.
>
>
> > gtalk is sort of missing key telephony features like the concept of pdd
> and
> > indication progress etc.
> >
> > Try answering the call in your dialplan before you bridge.
> >
> > I think the gtalk client is getting impatient because you did not answer
> > soon enough with an ip choice
> > so it starts sending all these extra ones at you and gave up on the one
> we
> > chose.
> >
> >
>
> I tested on another server runing FreeSWITCH Version 1.0.trunk
> (16958M), it works.
>
> see http://pastebin.freeswitch.org/12463
>
> interesting, messages like the same, but I see the working version has
> one more line on line 174:
>
> http://pastebin.freeswitch.org/12463 line: 173 to 175
> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
> [dingaling/gmail.com/seven at idapted.com] has been answered
> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> CS_ROUTING
>
>
> http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)
>
> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> CS_ROUTING
>
> Do you see any other difference? Thanks.
>
>
> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > FS would prefer you run your kernel at 1000hz you are probably on a VM
> >> > if
> >> > its 10000hz
> >> >
> >>
> >> It's a  physical machine. And I'm also running 20 skype clients. as
> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> >> 250 HZ. weird.
> >>
> >>
> >> here is the dingaling log with dl_debug on
> >>
> >> http://pastebin.freeswitch.org/12453
> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
> >>
> >>
> >> still get this message on client:
> >>
> >> Sorry! The voice chat with language_lesson failed because of a network
> >> problem at 5:39 PM. Please try again.
> >>  Click here to report this to Google
> >>
> >>
> >>
> >> Thanks.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> >
> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> >> Windows client and web site on Mac. They both work on the other two
> >> >> instances of FreeSWITCH I mentioned in this thread.
> >> >>
> >> >> btw, can you give me a hint on the timer warning?
> >> >>
> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> > you would have to execute
> >> >> >
> >> >> > dl_debug on
> >> >> >
> >> >> > to get a trace that would tell us anything about that?
> >> >> >
> >> >> > Are you using the windows gtalk client or the web site?
> >> >> > We are following google who have been pushing for the online client
> >> >> > embedded
> >> >> > in the gmail website.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> here is the client:
> >> >> >>
> >> >> >> <profile type="client">
> >> >> >>  <param name="name" value="gmail.com"/>
> >> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >> >>  <param name="password" value="password"/>
> >> >> >>  <param name="server" value="talk.google.com"/>
> >> >> >>  <param name="dialplan" value="XML"/>
> >> >> >>  <param name="contex" value="public"/>
> >> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >> >>  <param name="rtp-ip" value="auto"/>
> >> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >> >>  <param name="auto-login" value="true"/>
> >> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >> >>  <param name="sasl" value="plain"/>
> >> >> >>  <param name="tls" value="true"/>
> >> >> >>  <param name="exten" value="5000"/>
> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
> >> >> >> </profile>
> >> >> >>
> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> >> > ALL,
> >> >> >> >
> >> >> >> > I just upgrade FS to 16992.
> >> >> >> >
> >> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
> >> >> >> > sounds
> >> >> >> > ok.
> >> >> >> >
> >> >> >> >
> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
> microseconds
> >> >> >> > detected!
> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
> >> >> >> > experience audio problems.
> >> >> >> >
> >> >> >> > uname -a
> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
> 08:27:05
> >> >> >> > UTC
> >> >> >> > 2008 i686 GNU/Linux
> >> >> >> >
> >> >> >> >
> >> >> >> > I used a script to check the interupts
> >> >> >> >
> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >> >
> >> >> >> >
> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> >> >> >> > answer.
> >> >> >> > I
> >> >> >> > originate a call from FS, the console blocks, gtalk client ring
> >> >> >> > and
> >> >> >> > it
> >> >> >> > looks ok when I hit answer, but no sound, FS console still
> blocks.
> >> >> >> > a
> >> >> >> > few seconds later gtalk client shows:
> >> >> >> >
> >> >> >> > Sorry! The voice chat with language_lesson failed because of a
> >> >> >> > network
> >> >> >> > problem at 5:50 PM. Please try again.
> >> >> >> >  Click here to report this to Google.
> >> >> >> >
> >> >> >> >
> >> >> >> > here is a log with bgapi:
> >> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >> >
> >> >> >> >
> >> >> >> > rev 14696 works on the same server same conf. another server
> >> >> >> > running
> >> >> >> > 16958M works at the same time. I have no clue to check, is it
> >> >> >> > related
> >> >> >> > to the timer as in 1)? can some help me to take a look? let me
> >> >> >> > know
> >> >> >> > if
> >> >> >> > you need "dl_debug on" logs.
> >> >> >> >
> >> >> >> > Thanks.
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From ivdreg at gmail.com  Wed Mar 17 07:33:49 2010
From: ivdreg at gmail.com (ivdreg ivdreg)
Date: Wed, 17 Mar 2010 16:33:49 +0200
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
	<b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
Message-ID: <b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>

Jira opened. It's for my first time so may be not perfect.

2010/3/17 Rupa Schomaker <rupa at rupa.com>

> That should work.  Can you open a jira against mod_limit and assign it to
> me?  Thanks.
>
>
> On Wed, Mar 17, 2010 at 5:42 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> Thanks again Rupa,
>>
>> Is it possible to have more than one limit per call ? I have an issue in
>> this case with calls per period limiting.
>>
>> This is part of my dial-plan before bridge:
>>
>> <action application="limit_hash" data="caller ${destination_number} 5
>> !INCOMING_CALL_BARRED" />
>> <action application="limit_hash" data="crcd_${destination_number}
>> ${caller_id_number} 2 !OUTGOING_CALL_BARRED" />
>> <action application="limit_hash" data="cr_${sip_h_X-SRC-Oper}
>> ${caller_id_number} 20/86400 !OUTGOING_CALL_BARRED" />
>>
>> These are vars for 3 calls (not more than 10 sec between them):
>>
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>> --
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>> --
>> variable_limit_usage_caller_111111111111: [1]
>> variable_limit_rate_caller_111111111111: [0]
>> variable_limit_usage_crcd_111111111111_99999999999: [1]
>> variable_limit_rate_crcd_111111111111_99999999999: [0]
>> variable_limit_usage: [1]
>> variable_limit_usage_cr_TEST_99999999999: [1]
>> variable_limit_rate: [1]
>> variable_limit_rate_cr_TEST_99999999999: [1]
>>
>> I think that "variable_limit_usage_cr_TEST_99999999999" must be 3 in this
>> case. Is it true ? Does limit_usage and limit_rate that have no realm need
>> to exist also ?
>>
>> Thanks
>>
>>
>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>
>>> limit and limit_hash should work the same with regards to disconnect
>>> cause.  If you see a difference, let me know.
>>>
>>>
>>> On Tue, Mar 16, 2010 at 1:58 PM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>>>
>>>> Thanks Rupa,
>>>>
>>>> This was good clarification. I want to use mod_limit via 4 servers but
>>>> because traffic is equally balanced between them I will use limit_hash and
>>>> calculate how many calls every server will serve.
>>>>
>>>> One more question ? Is it necessary limit to transfer to limit_exceeded
>>>> or there is a way to set directly disconnect cause like in limit_hash ?
>>>>
>>>> Cheers
>>>>
>>>> 2010/3/16 Rupa Schomaker <rupa at rupa.com>
>>>>
>>>> 1) limit_hash use an in-memory hash for it's storage. There is nor
>>>>> storing to a database when using limit_hash.  If you want to use the
>>>>> database, then use plain limit.
>>>>>
>>>>> 2) If you want a max calls in a 24hrs period use you rate with
>>>>> limit_hash.  This should be about a true max per period, not concurrent.
>>>>>
>>>>> limit_hash realm id 50/86400
>>>>>
>>>>> should do what you want -- from memory.  double check the wiki/help to
>>>>> verify syntax.
>>>>>
>>>>> On Tue, Mar 16, 2010 at 10:20 AM, ivdreg ivdreg <ivdreg at gmail.com>wrote:
>>>>>
>>>>>> Hi All,
>>>>>>
>>>>>> I have some questions regarding mod_limit.
>>>>>>
>>>>>> 1. I'm using 16674 in production but tested and with 17001. I try
>>>>>> mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using
>>>>>> limit_hash never tries to write data to SQL. It works (counts sessions) on
>>>>>> log screen but no data in MySQL/Sqlite. It seems that data is cached but
>>>>>> cache never flushed to DB. Is anyone has this problem. My limits are:
>>>>>>
>>>>>> <action application="limit_hash" data="op_${sip_h_X-SRC-Cust}
>>>>>> ${destination_number} 5 !INCOMING_CALL_BARRED" />
>>>>>> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust}
>>>>>> ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" />
>>>>>>
>>>>>> and they are applied on inbound call profile in same extension.
>>>>>>
>>>>>>
>>>>>> 2. I need to implement rule maximum of 50 calls per day/24h from the
>>>>>> same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule
>>>>>> in example "Limit calls to 5 calls every 10 minutes" is about concurrent
>>>>>> call but not calls processed as count. Any idea ?
>>>>>>
>>>>>> Thanks for any help
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -Rupa
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 07:41:05 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 17 Mar 2010 07:41:05 -0700
Subject: [Freeswitch-users] RFC3578
In-Reply-To: <241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
References: <241d382f1003112320t173994d1gbb9b0e26ff2ba7ff@mail.gmail.com>
	<B280698A-4969-4FA3-8A40-DD28A81CCC21@freeswitch.org>
	<241d382f1003170132g111ab695p809f2617ec593453@mail.gmail.com>
Message-ID: <147345D8-6862-4B71-AF6A-27D0C686A78D@freeswitch.org>

Does the Huawei even claim to support rfc3578? If so you have proof  
that they are not following the spec. Contact them to find out what's  
going on. I'd love to see what happens if you ask them for a refund...
-MC

Sent from my iPhone

On Mar 17, 2010, at 1:32 AM, Timur Irmatov <irmatov at gmail.com> wrote:

> On Fri, Mar 12, 2010 at 5:56 PM, Brian West <brian at freeswitch.org>  
> wrote:
>> It already works use the respond application in the dialplan to  
>> respond 484
>
> Thanks for the hint, Brian. I have a SoftX3000 softswitch sending
> calls to FreeSWITCH. SoftX3000 is configured to send anything between
> 5 and 20 digits. I want FreeSWITCH to respond 484 until it receives
> enough digits to make a call.
>
> Here is an excerpt from my dialplan:
>
> <extension name="public_extensions">
>  <condition field="destination_number" expression="^006[0-9]{0,13}$">
>    <action application="respond" data="484"/>
>  </condition>
> </extension>
>
> In the first INVITE, softswitch sends 5 digits: 00610. FreeSWITCH
> answers 484 just as I want it to. Softswitch sends another INVITE,
> this time with 6 digits: 006107, but *without* SDP. Missing SDP upsets
> FreeSWITCH and it responds 480, call finishes. Here is log from
> FreeSWITCH:
>
> 2010-03-17 09:45:07.329047 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/1220139 at 10.0.2.5 [dc6f5d24-317f-11df-9775-c1f542b963a5]
> 2010-03-17 09:45:07.331047 [INFO] mod_dialplan_xml.c:418 Processing
> 1220139->00610 in context public
> 2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:185
> sofia/external/1220139 at 10.0.2.5 has executed the last dialplan
> instruction, hanging up.
> 2010-03-17 09:45:07.331047 [NOTICE] switch_core_state_machine.c:187
> Hangup sofia/external/1220139 at 10.0.2.5 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1179 Session
> 3 (sofia/external/1220139 at 10.0.2.5) Ended
> 2010-03-17 09:45:07.332066 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
> 2010-03-17 09:45:08.448839 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/1220139 at 10.0.2.5 [dd1a3776-317f-11df-9776-c1f542b963a5]
> 2010-03-17 09:45:08.448839 [INFO] sofia.c:4388 No SDP in INVITE and
> 3pcc not enabled, hanging up.
> 2010-03-17 09:45:08.449840 [NOTICE] sofia.c:4390 Hangup
> sofia/external/1220139 at 10.0.2.5 [CS_NEW] [MANDATORY_IE_MISSING]
> 2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1179 Session
> 4 (sofia/external/1220139 at 10.0.2.5) Ended
> 2010-03-17 09:45:08.450840 [NOTICE] switch_core_session.c:1181 Close
> Channel sofia/external/1220139 at 10.0.2.5 [CS_DESTROY]
>
>
> And here are SIP messages, capture made with ngrep:
>
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> INVITE sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>.
> CSeq: 1 INVITE.
> Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
> Supported: 100rel.
> P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
> User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
> Max-Forwards: 70.
> Allow:  
> INVITE, 
> ACK, 
> CANCEL, 
> OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
> Content-Length: 263.
> Content-Type: application/sdp.
> .
> v=0.
> o=HuaweiSoftX3000 98044 98044 IN IP4 10.0.2.5.
> s=Sip Call.
> c=IN IP4 172.16.12.1.
> t=0 0.
> m=audio 50000 RTP/AVP 8 0 18 4 2.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:2 G726-32/8000.
> a=fmtp:18 annexb=yes.
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 484 Address Incomplete.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> CSeq: 1 INVITE.
> Contact: <sip:00610 at 172.16.12.11:5090;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Length: 0.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> ACK sip:00610 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bK356eaa89a.
> Call-ID: a46545ffe2c776583bb02de315e8761a at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=c5514b7f.
> To: <sip:00610 at 172.16.12.11;user=phone>;tag=4gHUF5NN21SXS.
> CSeq: 1 ACK.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> INVITE sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>.
> CSeq: 1 INVITE.
> Contact: <sip:1220139 at 10.0.2.5:5070;user=phone>.
> P-Asserted-Identity: <sip:1220139 at 10.0.2.5;user=phone>.
> User-Agent: Huawei SoftX3000 V300R006B06D060SP32.
> Max-Forwards: 70.
> Allow:  
> INVITE, 
> ACK, 
> CANCEL, 
> OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Content-Length: 0.
> .
>
> #
> U 172.16.12.11:5090 -> 10.0.2.5:5070
> SIP/2.0 480 Temporarily Unavailable.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> CSeq: 1 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING".
> Content-Length: 0.
> P-Asserted-Identity: "006107" <sip:006107 at 172.16.12.11>.
> .
>
> #
> U 10.0.2.5:5070 -> 172.16.12.11:5090
> ACK sip:006107 at 172.16.12.11:5090;user=phone SIP/2.0.
> Via: SIP/2.0/UDP 10.0.2.5:5070;branch=z9hG4bKd3fe1aa88.
> Call-ID: 07f13e02d6d0be530d5da938452260a0 at 10.0.2.5.
> From: <sip:1220139 at 10.0.2.5;user=phone>;tag=df50ba93.
> To: <sip:006107 at 172.16.12.11;user=phone>;tag=5SamH06rZaggN.
> CSeq: 1 ACK.
> Max-Forwards: 70.
> Content-Length: 0.
>
> RFC3578 says:
>
> "The new INVITE has the same Call-ID and the same From header field
> including the tag as the first INVITE sent, but has an updated
> Request-URI."
>
> SoftX3000 sends second INVITE with different Call-ID. Is there
> anything I can do on FreeSWITCH to solve this problem and continue
> receiving additional digits?
>
>
> -- 
> Timur Irmatov, xmpp:irmatov at jabber.ru
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From tculjaga at gmail.com  Wed Mar 17 07:46:57 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 17 Mar 2010 15:46:57 +0100
Subject: [Freeswitch-users] Release Cause in response message
Message-ID: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>

hello folks,


anyone knows how to pass a Release cause/ Disconnect Reason when firing
respond from a dialplan ... e.g.:

Ive tried with sip_h_Reason=Q.850;cause=27 but it is not working :)



   <extension name="parseResult">
      <condition field="destination_number" expression="^parseResult$"/>
      <condition field="${componentStatus}" expression="^1$"
break="on-true">
         <action application="log" data="INFO ######################## CALL
FAILED REASON 1' ##########\n"/>

         <action application="set" data="sip_h_Reason=Q.850;cause=16"/>

         <action application="respond" data="480 Temporarily Unavailable"/>
      </condition>
   </extension>


any hint?
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From delorenzodesign at gmail.com  Wed Mar 17 07:52:16 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Wed, 17 Mar 2010 10:52:16 -0400
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
	<5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
Message-ID: <5c9dcbfb1003170752rd35a064s9457006ab17630cb@mail.gmail.com>

Ok, so I updated my install to include OpenZap.  It looks like the keys are
being detected, but not being returned by playAndGetDigits.

2010-03-17 10:49:21.207160 [DEBUG] sofia.c:4152 Channel
sofia/external/19736327407 entering state [ready][200]
2010-03-17 10:49:21.208170 [NOTICE] sofia.c:4676 Channel
[sofia/external/19736327407] has been answered
2010-03-17 10:49:21.208170 [DEBUG] switch_ivr_originate.c:3140 Originate
Resulted in Success: [sofia/external/19736327407]
2010-03-17 10:49:21.208170 [DEBUG] switch_cpp.cpp:497
(sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
CS_SOFT_EXECUTE
2010-03-17 10:49:21.208170 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:21.209171 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:200
sofia/external/19736327407 Standard SOFT_EXECUTE
2010-03-17 10:49:21.209171 [DEBUG] switch_core_state_machine.c:354
(sofia/external/19736327407) State SOFT_EXECUTE going to sleep
2010-03-17 10:49:21.241184 [DEBUG] switch_rtp.c:2064 Correct ip/port
confirmed.
*2010-03-17 10:49:25.582703 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:1440*
2010-03-17 10:49:25.582703 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
*2010-03-17 10:49:27.342662 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:480*
*2010-03-17 10:49:28.042790 [DEBUG] switch_rtp.c:2426 RTP RECV DTMF 1:48*0
2010-03-17 10:49:31.829092 [NOTICE] sofia.c:481 Hangup
sofia/external/19736327407 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2010-03-17 10:49:31.829092 [DEBUG] switch_channel.c:2071 Send signal
sofia/external/19736327407 [KILL]
2010-03-17 10:49:31.829092 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.829092 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_HANGUP
2010-03-17 10:49:31.830229 [DEBUG] switch_core_state_machine.c:499
(sofia/external/19736327407) State HANGUP
2010-03-17 10:49:31.830229 [DEBUG] mod_sofia.c:405
sofia/external/19736327407 Overriding SIP cause 480 with 200 from the other
leg
2010-03-17 10:49:31.830229 [DEBUG] mod_sofia.c:411 Channel
sofia/external/19736327407 hanging up, cause: NORMAL_CLEARING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:46
sofia/external/19736327407 Standard HANGUP, cause: NORMAL_CLEARING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:499
(sofia/external/19736327407) State HANGUP going to sleep
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:333
(sofia/external/19736327407) State Change CS_HANGUP -> CS_REPORTING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:314
(sofia/external/19736327407) Running State Change CS_REPORTING
2010-03-17 10:49:31.831230 [DEBUG] switch_core_state_machine.c:590
(sofia/external/19736327407) State REPORTING
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:53
sofia/external/19736327407 Standard REPORTING, cause: NORMAL_CLEARING
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:590
(sofia/external/19736327407) State REPORTING going to sleep
2010-03-17 10:49:31.832242 [DEBUG] switch_core_state_machine.c:327
(sofia/external/19736327407) State Change CS_REPORTING -> CS_DESTROY
2010-03-17 10:49:31.832242 [DEBUG] switch_core_session.c:1018 Send signal
sofia/external/19736327407 [BREAK]
2010-03-17 10:49:31.832242 [DEBUG] switch_core_session.c:1161 Session 1
(sofia/external/19736327407) Locked, Waiting on external entities
*2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 Digits received
during call from playAndGetDigits:*
2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 Did this check out? no
2010-03-17 10:49:31.841241 [INFO] switch_cpp.cpp:1142 We need to try this
callee again.



On Wed, Mar 17, 2010 at 10:00 AM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> How do I do that in Lua?  session:execute("dtmf","start") doesn't work.
>
>
> On Tue, Mar 16, 2010 at 11:38 PM, Michael S Collins <msc at freeswitch.org>wrote:
>
>> Perhaps the DTMFs are coming inband. Try the start_dtmf dialplan app.
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 16, 2010, at 7:05 PM, Michael De Lorenzo <
>> delorenzodesign at gmail.com> wrote:
>>
>> So I'm making calls using VoiceNetwork, are you suggesting there's a
>> problem there?  Or on the line where the call is received (I've tried on a
>> couple of different networks/providers)?
>>
>> On Tue, Mar 16, 2010 at 9:34 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> You inbound DID Provider
>>>
>>>
>>>
>>> On Tue, Mar 16, 2010 at 9:16 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>> delorenzodesign at gmail.com> wrote:
>>>
>>>> What do you mean about the provider?  I'm not sure I understand that
>>>> part.  I'm trying to call a number and have the call's recipient type the
>>>> digits in.
>>>>
>>>>
>>>> On Tue, Mar 16, 2010 at 9:07 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>> lloyd.aloysius at gmail.com> wrote:
>>>>
>>>>> here is the test script I used
>>>>>
>>>>> session:answer();
>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>> "enter-digits.wav", "", "\\d+");
>>>>>
>>>>> Here is the console log
>>>>>
>>>>> 2010-03-17 01:04:50.374761 [DEBUG] switch_ivr_play_say.c:1444 done
>>>>> playing file
>>>>> *2010-03-17 01:04:50.874736 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 1:800*
>>>>> *2010-03-17 01:04:51.104725 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 2:640*
>>>>> *2010-03-17 01:04:51.414710 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 3:800*
>>>>> *2010-03-17 01:04:51.774692 [DEBUG] switch_rtp.c:2417 RTP RECV DTMF
>>>>> 4:640*
>>>>> 2010-03-17 01:04:51.774692 [DEBUG] switch_ivr_play_say.c:1737 Test
>>>>> Regex [1234][\d+]
>>>>>
>>>>> You may have some issue with the provider. Check using a local
>>>>> extension.
>>>>>
>>>>>
>>>>> On Tue, Mar 16, 2010 at 8:51 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>> delorenzodesign at gmail.com> wrote:
>>>>>
>>>>>> Ok, I updated the console log level, and there are no DTMF values
>>>>>> being detected:
>>>>>>
>>>>>> 4268 -> 74.51.38.156 port 15150 codec: 0 ms: 20
>>>>>> 2010-03-16 20:49:06.534030 [DEBUG] switch_rtp.c:1181 Starting timer
>>>>>> [soft] 160 bytes per 20ms
>>>>>> 2010-03-16 20:49:06.535030 [NOTICE] sofia_glue.c:3128 Pre-Answer
>>>>>> sofia/external/19736327407!
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>>> sofia/external/19736327407 entering state [completing][200]
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4150 Remote SDP:
>>>>>> v=0
>>>>>> o=root 21648 21649 IN IP4 74.51.38.156
>>>>>> s=session
>>>>>> c=IN IP4 74.51.38.156
>>>>>> t=0 0
>>>>>> m=audio 15150 RTP/AVP 0 101
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=silenceSupp:off - - - -
>>>>>> a=ptime:20
>>>>>>
>>>>>> 2010-03-16 20:49:13.762610 [DEBUG] sofia.c:4139 Channel
>>>>>> sofia/external/19736327407 entering state [ready][200]
>>>>>> 2010-03-16 20:49:13.763613 [NOTICE] sofia.c:4663 Channel
>>>>>> [sofia/external/19736327407] has been answered
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_ivr_originate.c:3107
>>>>>> Originate Resulted in Success: [sofia/external/19736327407]
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_cpp.cpp:497
>>>>>> (sofia/external/19736327407) State Change CS_CONSUME_MEDIA ->
>>>>>> CS_SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.763613 [DEBUG] switch_core_session.c:1018 Send
>>>>>> signal sofia/external/19736327407 [BREAK]
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_ivr_play_say.c:1178 Codec
>>>>>> Activated L16 at 8000hz 1 channels 20ms
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:314
>>>>>> (sofia/external/19736327407) Running State Change CS_SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>>> (sofia/external/19736327407) State SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] mod_sofia.c:510 SOFIA SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:200
>>>>>> sofia/external/19736327407 Standard SOFT_EXECUTE
>>>>>> 2010-03-16 20:49:13.764618 [DEBUG] switch_core_state_machine.c:354
>>>>>> (sofia/external/19736327407) State SOFT_EXECUTE going to sleep
>>>>>> 2010-03-16 20:49:13.794735 [DEBUG] switch_rtp.c:2055 Correct ip/port
>>>>>> confirmed.
>>>>>> 2010-03-16 20:49:25.055402 [DEBUG] switch_ivr_play_say.c:1470 done
>>>>>> playing file
>>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Digits received
>>>>>> during call from playAndGetDigits:
>>>>>> 2010-03-16 20:49:30.074803 [INFO] switch_cpp.cpp:1142 Did this check
>>>>>> out? no
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 16, 2010 at 8:37 PM, Aloysius Lloyd <<lloyd.aloysius at gmail.com>
>>>>>> lloyd.aloysius at gmail.com> wrote:
>>>>>>
>>>>>>> set the console log level to 7. Then you can see the DTMF values in
>>>>>>> console log , while you are pressing the keys.
>>>>>>>
>>>>>>> Lloyd
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Mar 16, 2010 at 8:23 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>
>>>>>>>> So I tried Michael's suggestion of:
>>>>>>>>
>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>
>>>>>>>> but it still didn't detect that I had pressed any keys.  What would
>>>>>>>> cause this?  Is there someplace I can check to see if anything at all is
>>>>>>>> being detected?
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Mar 16, 2010 at 5:00 PM, Michael Collins <<msc at freeswitch.org>
>>>>>>>> msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Mar 16, 2010 at 1:21 PM, Michael De Lorenzo <<delorenzodesign at gmail.com>
>>>>>>>>> delorenzodesign at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I'm having some trouble getting consistent results with
>>>>>>>>>> playAndGetDigits, sometimes the digits are record and other times they're
>>>>>>>>>> not.  It seemed to have something to do with how quickly the digits were
>>>>>>>>>> pressed, but now that no longer seems to be the case.
>>>>>>>>>>
>>>>>>>>>> I'm trying to have users press a confirmation code, in this case
>>>>>>>>>> "1111" although I'd like to make it accept "1" or "11" or "111" or "1111" if
>>>>>>>>>> possible.
>>>>>>>>>>
>>>>>>>>>> Can anyone point me in the right direction?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>     session =
>>>>>>>>>> freeswitch.Session("{ignore_early_media=true}sofia/gateway/" .. provider ..
>>>>>>>>>> "/1" .. number_to_call);
>>>>>>>>>>
>>>>>>>>>>     while(session:ready()) do
>>>>>>>>>>        session:answer();
>>>>>>>>>>        session:setInputCallback("onInput","true");
>>>>>>>>>>
>>>>>>>>>>        --playbackMessage(check_message);
>>>>>>>>>>        digits = session:playAndGetDigits(4, 4, 1, 5000, "",
>>>>>>>>>> get_digits_message, "", "[1]");
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> You are specifying a minimum of four and a maximum of four digits.
>>>>>>>>> Also, your regex will match any string that has a digit 1 in it, which may
>>>>>>>>> or may not be what you are looking for. Lastly, you haven't specified a
>>>>>>>>> terminator key (like "#") but you are giving the user only one try and
>>>>>>>>> waiting 5000ms for the entry. Try something like this:
>>>>>>>>>
>>>>>>>>> digits = session:playAndGetDigits(1, 4, 1, 5000, "#",
>>>>>>>>> get_digits_message, "", "\\d+");
>>>>>>>>>
>>>>>>>>> Let us know if that works. Oh, one other thing: you might want to
>>>>>>>>> add an invalid message because playAndGetDigits will handle invalid input
>>>>>>>>> for you.
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>> P.S. - This topic is covered in greater detail in Chapter 7 of the
>>>>>>>>> upcoming FreeSWITCH book. ;)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>        freeswitch.consoleLog("info", digits .. "\n");
>>>>>>>>>>        freeswitch.consoleLog("info", "Did this check out? " ..
>>>>>>>>>> (check_message == true and "yes" or "no" .. "\n"));
>>>>>>>>>>
>>>>>>>>>>        -- we got what we're looking for, so we can end this
>>>>>>>>>>        if(check_success) then
>>>>>>>>>>           freeswitch.consoleLog("info", "A positive response was
>>>>>>>>>> received from this check.\n");
>>>>>>>>>>           break;
>>>>>>>>>>        else
>>>>>>>>>>           -- not a positive response
>>>>>>>>>>           -- if we reached our max attempts, we're finished move
>>>>>>>>>> on
>>>>>>>>>>           if(attempts == max_attempts) then
>>>>>>>>>>             freeswitch.consoleLog("info", "We've reached our
>>>>>>>>>> maximum attempts for this number.\n");
>>>>>>>>>>             break;
>>>>>>>>>>           end
>>>>>>>>>>        end
>>>>>>>>>>     end
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From dujinfang at gmail.com  Wed Mar 17 08:03:12 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 17 Mar 2010 23:03:12 +0800
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
	<191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
Message-ID: <23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>

No.

FreeSWITCH runs on public IP.  my gtalk client behind NAT.  As I said,
old versions(e.g. 14696) work on the same server with the same config.

2010/3/17 Anthony Minessale <anthony.minessale at gmail.com>:
> Were you using gtalk and FS both behind nat with the public gtalk server by
> any chance?
> This setup requires special configuration.
>
>
> On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> > Are you bridging to sip or some ivr and making dingaling wait until the
>> > call
>> > answers?
>> >
>>
>> no I was originate dingaling/... &echo, ? also tried originate
>> dingaling/... ?extension XML public (which will hit dialplan and
>> answer() after client answers), neither work.
>>
>>
>> > gtalk is sort of missing key telephony features like the concept of pdd
>> > and
>> > indication progress etc.
>> >
>> > Try answering the call in your dialplan before you bridge.
>> >
>> > I think the gtalk client is getting impatient because you did not answer
>> > soon enough with an ip choice
>> > so it starts sending all these extra ones at you and gave up on the one
>> > we
>> > chose.
>> >
>> >
>>
>> I tested on another server runing FreeSWITCH Version 1.0.trunk
>> (16958M), it works.
>>
>> see http://pastebin.freeswitch.org/12463
>>
>> interesting, messages like the same, but I see the working version has
>> one more line on line 174:
>>
>> http://pastebin.freeswitch.org/12463 line: 173 to 175
>> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
>> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
>> [dingaling/gmail.com/seven at idapted.com] has been answered
>> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
>> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
>> CS_ROUTING
>>
>>
>> http://pastebin.freeswitch.org/12453 ?line 217 - 218 ?(rev16992M)
>>
>> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
>> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
>> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
>> CS_ROUTING
>>
>> Do you see any other difference? Thanks.
>>
>>
>> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com> wrote:
>> >>
>> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> > FS would prefer you run your kernel at 1000hz you are probably on a
>> >> > VM
>> >> > if
>> >> > its 10000hz
>> >> >
>> >>
>> >> It's a ?physical machine. And I'm also running 20 skype clients. as
>> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
>> >> 250 HZ. weird.
>> >>
>> >>
>> >> here is the dingaling log with dl_debug on
>> >>
>> >> http://pastebin.freeswitch.org/12453
>> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
>> >>
>> >>
>> >> still get this message on client:
>> >>
>> >> Sorry! The voice chat with language_lesson failed because of a network
>> >> problem at 5:39 PM. Please try again.
>> >> ?Click here to report this to Google
>> >>
>> >>
>> >>
>> >> Thanks.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> >
>> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
>> >> >> Windows client and web site on Mac. They both work on the other two
>> >> >> instances of FreeSWITCH I mentioned in this thread.
>> >> >>
>> >> >> btw, can you give me a hint on the timer warning?
>> >> >>
>> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
>> >> >> > you would have to execute
>> >> >> >
>> >> >> > dl_debug on
>> >> >> >
>> >> >> > to get a trace that would tell us anything about that?
>> >> >> >
>> >> >> > Are you using the windows gtalk client or the web site?
>> >> >> > We are following google who have been pushing for the online
>> >> >> > client
>> >> >> > embedded
>> >> >> > in the gmail website.
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> here is the client:
>> >> >> >>
>> >> >> >> <profile type="client">
>> >> >> >> ?<param name="name" value="gmail.com"/>
>> >> >> >> ?<param name="login" value="xx at gmail.com/talk"/>
>> >> >> >> ?<param name="password" value="password"/>
>> >> >> >> ?<param name="server" value="talk.google.com"/>
>> >> >> >> ?<param name="dialplan" value="XML"/>
>> >> >> >> ?<param name="contex" value="public"/>
>> >> >> >> ?<param name="message" value="FreeSWITCH"/>
>> >> >> >> ?<param name="rtp-ip" value="auto"/>
>> >> >> >> ?<!-- if you are behind NAT, you should probably use stun -->
>> >> >> >> ?<!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
>> >> >> >> ?<param name="auto-login" value="true"/>
>> >> >> >> ?<param name="auto-reply" value="Welcome"/>
>> >> >> >> ?<param name="sasl" value="plain"/>
>> >> >> >> ?<param name="tls" value="true"/>
>> >> >> >> ?<param name="exten" value="5000"/>
>> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/> -->
>> >> >> >> </profile>
>> >> >> >>
>> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
>> >> >> >> > ALL,
>> >> >> >> >
>> >> >> >> > I just upgrade FS to 16992.
>> >> >> >> >
>> >> >> >> > 1) timer warning: How's possible the 10000 microseconds? audio
>> >> >> >> > sounds
>> >> >> >> > ok.
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
>> >> >> >> > microseconds
>> >> >> >> > detected!
>> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You may
>> >> >> >> > experience audio problems.
>> >> >> >> >
>> >> >> >> > uname -a
>> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
>> >> >> >> > 08:27:05
>> >> >> >> > UTC
>> >> >> >> > 2008 i686 GNU/Linux
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > I used a script to check the interupts
>> >> >> >> >
>> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
>> >> >> >> > LOC: ? 14892854 ? 14892843 ? 14892841 ? 14892830
>> >> >> >> > LOC: ? 14892955 ? 14892944 ? 14892942 ? 14892931
>> >> >> >> > LOC: ? 14893056 ? 14893045 ? 14893043 ? 14893032
>> >> >> >> > LOC: ? 14893157 ? 14893146 ? 14893144 ? 14893133
>> >> >> >> > LOC: ? 14893258 ? 14893247 ? 14893245 ? 14893234
>> >> >> >> > LOC: ? 14893359 ? 14893348 ? 14893346 ? 14893335
>> >> >> >> > LOC: ? 14893460 ? 14893449 ? 14893447 ? 14893436
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
>> >> >> >> > answer.
>> >> >> >> > I
>> >> >> >> > originate a call from FS, the console blocks, gtalk client ring
>> >> >> >> > and
>> >> >> >> > it
>> >> >> >> > looks ok when I hit answer, but no sound, FS console still
>> >> >> >> > blocks.
>> >> >> >> > a
>> >> >> >> > few seconds later gtalk client shows:
>> >> >> >> >
>> >> >> >> > Sorry! The voice chat with language_lesson failed because of a
>> >> >> >> > network
>> >> >> >> > problem at 5:50 PM. Please try again.
>> >> >> >> > ?Click here to report this to Google.
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > here is a log with bgapi:
>> >> >> >> > http://pastebin.freeswitch.org/12434
>> >> >> >> >
>> >> >> >> >
>> >> >> >> > rev 14696 works on the same server same conf. another server
>> >> >> >> > running
>> >> >> >> > 16958M works at the same time. I have no clue to check, is it
>> >> >> >> > related
>> >> >> >> > to the timer as in 1)? can some help me to take a look? let me
>> >> >> >> > know
>> >> >> >> > if
>> >> >> >> > you need "dl_debug on" logs.
>> >> >> >> >
>> >> >> >> > Thanks.
>> >> >> >> >
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > --
>> >> >> > Anthony Minessale II
>> >> >> >
>> >> >> > FreeSWITCH http://www.freeswitch.org/
>> >> >> > ClueCon http://www.cluecon.com/
>> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >> >
>> >> >> > AIM: anthm
>> >> >> > MSN:anthony_minessale at hotmail.com
>> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> >> > IRC: irc.freenode.net #freeswitch
>> >> >> >
>> >> >> > FreeSWITCH Developer Conference
>> >> >> > sip:888 at conference.freeswitch.org
>> >> >> > iax:guest at conference.freeswitch.org/888
>> >> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> >> > pstn:+19193869900
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> >
>> >> > --
>> >> > Anthony Minessale II
>> >> >
>> >> > FreeSWITCH http://www.freeswitch.org/
>> >> > ClueCon http://www.cluecon.com/
>> >> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >> >
>> >> > AIM: anthm
>> >> > MSN:anthony_minessale at hotmail.com
>> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> > IRC: irc.freenode.net #freeswitch
>> >> >
>> >> > FreeSWITCH Developer Conference
>> >> > sip:888 at conference.freeswitch.org
>> >> > iax:guest at conference.freeswitch.org/888
>> >> > googletalk:conf+888 at conference.freeswitch.org
>> >> > pstn:+19193869900
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From infos at madovsky.org  Wed Mar 17 08:05:21 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 11:05:21 -0400
Subject: [Freeswitch-users] freeswitch to cisco gateway
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705><ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
Message-ID: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>

So why not GPL or Gnu Affero public licence ?

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 17, 2010 9:23 AM
Subject: Re: [Freeswitch-users] freeswitch to cisco gateway


> This is where it gets hairy... Yes the MPL says you have to share your 
> modifications with the original author but as far as I can tell you don't 
> have to distribute to cause this clause to go into effect.
>
> Its like this:
>
> You change switch_ivr_originate.c, you HAVE to share those changes with 
> Anthony.
> You create your own mod_kewlshit.c, its yours to do with what you want and 
> you don't have to share.
>
> You change any code that is included with the download YOU MUST share with 
> the original author.
>
> /b
>
>
> On Mar 16, 2010, at 11:30 PM, jay binks wrote:
>
>> Im not on the FS dev team, but I think you may be a little mistaken.
>>
>> generally your free to make modifications to open source code ( for
>> internal use )
>> however its when you distribute the resultant product, then you have
>> to make the source available..
>>
>> if your only using it in-house, I dont believe your required to
>> contribute the code back.
>> ( however your going to end up having issues maintaining your patch
>> set over a long period.. like can be seen here )
>>
>> see here :
>>
>>> must be made available in Source Code form under the terms of this 
>>> License either on the
>>> same media as an Executable version
>>
>> if your not distributing your executable to any 3rd parties... then I
>> dont believe you have to distribute the modified source..
>>
>> Correct me if Im wrong though guys...
>>
>> Jay
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 




From rupa at rupa.com  Wed Mar 17 08:28:41 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 17 Mar 2010 10:28:41 -0500
Subject: [Freeswitch-users] Mod_limit questions
In-Reply-To: <b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>
References: <b437a0e1003160820k37b1c853n9f5e7eed4651fcd4@mail.gmail.com>
	<b39273531003160858o6311495boedf1b7d79bda5500@mail.gmail.com>
	<b437a0e1003161158q55b793dbn983d85f6bd7bff7b@mail.gmail.com>
	<b39273531003161241h7bc020a5n8579ab33d31f3470@mail.gmail.com>
	<b437a0e1003170342t74c22b27k59a760ab2bab2d6c@mail.gmail.com>
	<b39273531003170641m17fb3e7br5d8f3eded0f3ef43@mail.gmail.com>
	<b437a0e1003170733h44e1b5edm92516d79b4c92d7a@mail.gmail.com>
Message-ID: <b39273531003170828n32e2e2d6ic7170afcc09e6c1e@mail.gmail.com>

Thanks

On Wed, Mar 17, 2010 at 9:33 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:

> Jira opened. It's for my first time so may be not perfect.
>
> --
-Rupa
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Mar 17 08:31:44 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 10:31:44 -0500
Subject: [Freeswitch-users] A few questions after upgrade to trunk
In-Reply-To: <23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>
References: <23f91031003150308j5e5938fasd0cf9d2adb5147d5@mail.gmail.com>
	<23f91031003150327g182f0743r73c7002514471817@mail.gmail.com>
	<191c3a031003150928s7ec8a562hddff8a89bad4ef68@mail.gmail.com>
	<23f91031003151001t2a6dfcc5p2ed58b6c2376f899@mail.gmail.com>
	<191c3a031003151018l670fdab6t5fc638115bb119d@mail.gmail.com>
	<23f91031003160253n1a9ad426gaca1b13f4a2ee64c@mail.gmail.com>
	<191c3a031003160645n69a9ef8dr8271673a18bfc34a@mail.gmail.com>
	<23f91031003170301t20b97baaj1a3c81d4aa586a0b@mail.gmail.com>
	<191c3a031003170716y33f53955nfd71f92e2463fd36@mail.gmail.com>
	<23f91031003170803gad9550dxf36ce7cf3d8871d6@mail.gmail.com>
Message-ID: <191c3a031003170831l2a89137aw98df9d85c109c31@mail.gmail.com>

And trunk also works for us on 2 different tests so I am going to guess its
a configuration issue or some
other variable besides the version of the software.

mod_dingaling has not really changed in months.

We cannot reproduce it so if you want me to fix it you would have to let me
into your box when I have time.
our key is in the source tree in support-d/shinzon.pub

email me the login details at consulting at freeswitch.org and I'll have a
look.


On Wed, Mar 17, 2010 at 10:03 AM, Seven Du <dujinfang at gmail.com> wrote:

> No.
>
> FreeSWITCH runs on public IP.  my gtalk client behind NAT.  As I said,
> old versions(e.g. 14696) work on the same server with the same config.
>
> 2010/3/17 Anthony Minessale <anthony.minessale at gmail.com>:
> > Were you using gtalk and FS both behind nat with the public gtalk server
> by
> > any chance?
> > This setup requires special configuration.
> >
> >
> > On Wed, Mar 17, 2010 at 5:01 AM, Seven Du <dujinfang at gmail.com> wrote:
> >>
> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> > Are you bridging to sip or some ivr and making dingaling wait until
> the
> >> > call
> >> > answers?
> >> >
> >>
> >> no I was originate dingaling/... &echo,   also tried originate
> >> dingaling/...  extension XML public (which will hit dialplan and
> >> answer() after client answers), neither work.
> >>
> >>
> >> > gtalk is sort of missing key telephony features like the concept of
> pdd
> >> > and
> >> > indication progress etc.
> >> >
> >> > Try answering the call in your dialplan before you bridge.
> >> >
> >> > I think the gtalk client is getting impatient because you did not
> answer
> >> > soon enough with an ip choice
> >> > so it starts sending all these extra ones at you and gave up on the
> one
> >> > we
> >> > chose.
> >> >
> >> >
> >>
> >> I tested on another server runing FreeSWITCH Version 1.0.trunk
> >> (16958M), it works.
> >>
> >> see http://pastebin.freeswitch.org/12463
> >>
> >> interesting, messages like the same, but I see the working version has
> >> one more line on line 174:
> >>
> >> http://pastebin.freeswitch.org/12463 line: 173 to 175
> >> 2010-03-17 09:41:08.524787 [DEBUG] switch_rtp.c:1189 Not using a timer
> >> 2010-03-17 09:41:08.525781 [NOTICE] mod_dingaling.c:1156 Channel
> >> [dingaling/gmail.com/seven at idapted.com] has been answered
> >> 2010-03-17 09:41:08.525781 [DEBUG] mod_dingaling.c:1189
> >> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> >> CS_ROUTING
> >>
> >>
> >> http://pastebin.freeswitch.org/12453  line 217 - 218  (rev16992M)
> >>
> >> 2010-03-16 09:39:21.603723 [DEBUG] switch_rtp.c:1189 Not using a timer
> >> 2010-03-16 09:39:21.613769 [DEBUG] mod_dingaling.c:1189
> >> (dingaling/gmail.com/seven at idapted.com) State Change CS_INIT ->
> >> CS_ROUTING
> >>
> >> Do you see any other difference? Thanks.
> >>
> >>
> >> > On Tue, Mar 16, 2010 at 4:53 AM, Seven Du <dujinfang at gmail.com>
> wrote:
> >> >>
> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> > FS would prefer you run your kernel at 1000hz you are probably on a
> >> >> > VM
> >> >> > if
> >> >> > its 10000hz
> >> >> >
> >> >>
> >> >> It's a  physical machine. And I'm also running 20 skype clients. as
> >> >> 100HZ is recommended for skypiax. I think the kernel should be 100 or
> >> >> 250 HZ. weird.
> >> >>
> >> >>
> >> >> here is the dingaling log with dl_debug on
> >> >>
> >> >> http://pastebin.freeswitch.org/12453
> >> >> My public IP is 220.231.26.130, and nat to 192.168.1.27(me)
> >> >>
> >> >>
> >> >> still get this message on client:
> >> >>
> >> >> Sorry! The voice chat with language_lesson failed because of a
> network
> >> >> problem at 5:39 PM. Please try again.
> >> >>  Click here to report this to Google
> >> >>
> >> >>
> >> >>
> >> >> Thanks.
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> >
> >> >> > On Mon, Mar 15, 2010 at 11:01 AM, Seven Du <dujinfang at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> Thanks, It's late now, I will get logs tomorrow. I tested both on
> >> >> >> Windows client and web site on Mac. They both work on the other
> two
> >> >> >> instances of FreeSWITCH I mentioned in this thread.
> >> >> >>
> >> >> >> btw, can you give me a hint on the timer warning?
> >> >> >>
> >> >> >> 2010/3/16 Anthony Minessale <anthony.minessale at gmail.com>:
> >> >> >> > you would have to execute
> >> >> >> >
> >> >> >> > dl_debug on
> >> >> >> >
> >> >> >> > to get a trace that would tell us anything about that?
> >> >> >> >
> >> >> >> > Are you using the windows gtalk client or the web site?
> >> >> >> > We are following google who have been pushing for the online
> >> >> >> > client
> >> >> >> > embedded
> >> >> >> > in the gmail website.
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > On Mon, Mar 15, 2010 at 4:27 AM, Seven Du <dujinfang at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> here is the client:
> >> >> >> >>
> >> >> >> >> <profile type="client">
> >> >> >> >>  <param name="name" value="gmail.com"/>
> >> >> >> >>  <param name="login" value="xx at gmail.com/talk"/>
> >> >> >> >>  <param name="password" value="password"/>
> >> >> >> >>  <param name="server" value="talk.google.com"/>
> >> >> >> >>  <param name="dialplan" value="XML"/>
> >> >> >> >>  <param name="contex" value="public"/>
> >> >> >> >>  <param name="message" value="FreeSWITCH"/>
> >> >> >> >>  <param name="rtp-ip" value="auto"/>
> >> >> >> >>  <!-- if you are behind NAT, you should probably use stun -->
> >> >> >> >>  <!--<param name="ext-rtp-ip" value="stun:stun.xten.com"/> -->
> >> >> >> >>  <param name="auto-login" value="true"/>
> >> >> >> >>  <param name="auto-reply" value="Welcome"/>
> >> >> >> >>  <param name="sasl" value="plain"/>
> >> >> >> >>  <param name="tls" value="true"/>
> >> >> >> >>  <param name="exten" value="5000"/>
> >> >> >> >> <!-- <param name="local-network-acl" value="localnet.auto"/>
> -->
> >> >> >> >> </profile>
> >> >> >> >>
> >> >> >> >> 2010/3/15 Seven Du <dujinfang at gmail.com>:
> >> >> >> >> > ALL,
> >> >> >> >> >
> >> >> >> >> > I just upgrade FS to 16992.
> >> >> >> >> >
> >> >> >> >> > 1) timer warning: How's possible the 10000 microseconds?
> audio
> >> >> >> >> > sounds
> >> >> >> >> > ok.
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > [WARNING] switch_time.c:206 Timer resolution of 10000
> >> >> >> >> > microseconds
> >> >> >> >> > detected!
> >> >> >> >> > Do you have your kernel timer set to higher than 1 kHz? You
> may
> >> >> >> >> > experience audio problems.
> >> >> >> >> >
> >> >> >> >> > uname -a
> >> >> >> >> > Linux xxx.idapted.com 2.6.22-14-server #1 SMP Tue Feb 12
> >> >> >> >> > 08:27:05
> >> >> >> >> > UTC
> >> >> >> >> > 2008 i686 GNU/Linux
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > I used a script to check the interupts
> >> >> >> >> >
> >> >> >> >> > :~$ while :; do cat /proc/interrupts |grep LOC; sleep 1; done
> >> >> >> >> > LOC:   14892854   14892843   14892841   14892830
> >> >> >> >> > LOC:   14892955   14892944   14892942   14892931
> >> >> >> >> > LOC:   14893056   14893045   14893043   14893032
> >> >> >> >> > LOC:   14893157   14893146   14893144   14893133
> >> >> >> >> > LOC:   14893258   14893247   14893245   14893234
> >> >> >> >> > LOC:   14893359   14893348   14893346   14893335
> >> >> >> >> > LOC:   14893460   14893449   14893447   14893436
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > 2) run dingaling with client mode. dingaling doesn't pick up
> >> >> >> >> > answer.
> >> >> >> >> > I
> >> >> >> >> > originate a call from FS, the console blocks, gtalk client
> ring
> >> >> >> >> > and
> >> >> >> >> > it
> >> >> >> >> > looks ok when I hit answer, but no sound, FS console still
> >> >> >> >> > blocks.
> >> >> >> >> > a
> >> >> >> >> > few seconds later gtalk client shows:
> >> >> >> >> >
> >> >> >> >> > Sorry! The voice chat with language_lesson failed because of
> a
> >> >> >> >> > network
> >> >> >> >> > problem at 5:50 PM. Please try again.
> >> >> >> >> >  Click here to report this to Google.
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > here is a log with bgapi:
> >> >> >> >> > http://pastebin.freeswitch.org/12434
> >> >> >> >> >
> >> >> >> >> >
> >> >> >> >> > rev 14696 works on the same server same conf. another server
> >> >> >> >> > running
> >> >> >> >> > 16958M works at the same time. I have no clue to check, is it
> >> >> >> >> > related
> >> >> >> >> > to the timer as in 1)? can some help me to take a look? let
> me
> >> >> >> >> > know
> >> >> >> >> > if
> >> >> >> >> > you need "dl_debug on" logs.
> >> >> >> >> >
> >> >> >> >> > Thanks.
> >> >> >> >> >
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> >
> >> >> >> > --
> >> >> >> > Anthony Minessale II
> >> >> >> >
> >> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> >> > ClueCon http://www.cluecon.com/
> >> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >> >
> >> >> >> > AIM: anthm
> >> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >> >
> >> >> >> > FreeSWITCH Developer Conference
> >> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> >> > pstn:+19193869900
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> >
> >> >> > --
> >> >> > Anthony Minessale II
> >> >> >
> >> >> > FreeSWITCH http://www.freeswitch.org/
> >> >> > ClueCon http://www.cluecon.com/
> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >> >
> >> >> > AIM: anthm
> >> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> >> > IRC: irc.freenode.net #freeswitch
> >> >> >
> >> >> > FreeSWITCH Developer Conference
> >> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> >> > iax:guest at conference.freeswitch.org/888
> >> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> >> > pstn:+19193869900
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:+19193869900
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From fraserredmond at gmail.com  Tue Mar 16 15:21:45 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Tue, 16 Mar 2010 22:21:45 +0000
Subject: [Freeswitch-users] How long did the phone ring?
Message-ID: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>

I'm converting a call-center app from Asterisk to FreeSwitch (using xml and
javascript dialplans) and I think I've worked out how to do nearly
everything, except for tracking one important metric: How long the phone
rang before an agent picked it up.

I'm not even entirely sure what to call it in FreeSwitch lingo - so that
would be a good starting point :-)  (Media-duration maybe?)

I can work out the length of ringing for calls that didn't connect, by
logging when the bridge is attempted, and after it fails and comparing the
time. But if the person being called picks up, then all I know from that is
the total length of the call including the ringing/media time, not the media
time and connected time as 2 separate values

The info application gives several useful looking variables like:
Caller-Channel-Progress-Media-Time
Caller-Channel-Transfer-Time
...but they're still set to zero.

I thought I might be able to subtract  Caller-Channel-Created-Time  from
Caller-Channel-Answered-Time  to get the milliseconds of how long the phone
rang for - but there's no reliable correlation between the difference and
how long the phone rang.

In asterisk I achieved this by logging when I was about to make a call, then
running a macro when the Dial() connected, like so:
    exten => s,n,Macro(Dial_Attempted)
    exten => s,n,Dial(SIP/1001,20,M(Dial_Succeeded))
    exten => s,n,Macro(Dial_Failed)  ;Only gets here if the Dial didn't
connect.

Is there any way to do this in FreeSwitch?

Cheers,
Fraser
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From cucku.cucku at yahoo.com.vn  Wed Mar 17 08:29:17 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Wed, 17 Mar 2010 23:29:17 +0800 (SGT)
Subject: [Freeswitch-users] Newbie question on freeswitch
Message-ID: <130309.20630.qm@web76211.mail.sg1.yahoo.com>

HI list
i am new to Freeswitch, i am successfully install freeswitchand i use the default configi use 1000 and 1001 to register successfull and make call between 2 endpoints fine
after that, i create a new 8000.xml file which in the same location as 1000.xmlfreeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml<include>??<user id="8000">?? ?<params>?? ? ?<param name="password" value="$${default_password}"/>?? ? ?<param name="vm-password" value="1000"/>?? ?</params>?? ?<variables>?? ? ?<variable name="toll_allow" value="domestic,international,local"/>?? ? ?<variable name="accountcode" value="8000"/>?? ? ?<variable name="user_context" value="default"/>?? ? ?<variable name="effective_caller_id_name" value="Extension 8000"/>?? ? ?<variable name="effective_caller_id_number" value="8000"/>?? ? ?<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>?? ? ?<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>?? ? ?<variable name="callgroup" value="techsupport"/>?? ?</variables>??</user></include>
the 8000 is registered successfull but i cannot use the 1000 call to 8000
please helpHa`


      C?m gi?c an to?n h?n khi online. N?ng c?p l?n Internet Explorer 8 m?i, an to?n h?n,t?i ?u h?a cho Yahoo!. Ho?n to?n mi?n ph?. T?i IE8 t?i ??y! 
http://downloads.yahoo.com/vn/internetexplorer/
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From lloyd.aloysius at gmail.com  Wed Mar 17 08:43:55 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 11:43:55 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<8a19bf2e1003161743s1c12a965l82a0a516012663a@mail.gmail.com>
	<82dae1c01003161918j2e7a8307h4afaf8693ad479af@mail.gmail.com>
Message-ID: <8a19bf2e1003170843ub5bbec5pcc6773d2bbe8f646@mail.gmail.com>

Rupa & Tom , Thank you for the suggestion.

Lloyd

On Tue, Mar 16, 2010 at 10:18 PM, Tom Christensen <pavera at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_fifo#Dialplan_Example
> At least with mod_fifo (the call parking solution I'm using) there is a
> pretty clear example in the wiki.
>
> On Tue, Mar 16, 2010 at 6:43 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> when a user parked to a slot how to handle the park time out. Is there any
>> way to transfer to the call back to a receptionist.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> On Tue, Mar 16, 2010 at 5:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>>
>>>> Hello,
>>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>>> is really nice, straightforward, and I appreciate the design of the system.
>>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>>> really looking for something more stable, and I think I've found it here!
>>>>
>>>> One issue I've run into, and I'm not sure if its just that my use case
>>>> is abnormal, hence the email.
>>>>
>>>> I've worked with PBX systems for a number of years (11 now...), and in
>>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>>> feature.  By this I mean, you think party X (the intended recipient of the
>>>> call) is in the building, but they are not at their desk.  So, you park the
>>>> call, and then page, or by some other means attempt to locate party X and
>>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>>> party X or you can't find them in time, the call should time out back to the
>>>> parking party (receptionist) to be handled appropriately (message taken,
>>>> sent to voicemail, transferred to party X's boss)
>>>>
>>>> This scenario requires a few things:
>>>> 1) the receptionist/person who actually receives the call needs to be
>>>> able to put the call in a specific place (extension) that doesn't change
>>>> 2) Party X (the intended recipient) needs to be able to access that
>>>> specific place (extension) from any phone inside the company
>>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>>> party X and one for party Y, in that order, but party Y gets to a phone
>>>> first, they will be connected to party X's caller)
>>>> 4) The receptionist has to be protected from or provided with sufficient
>>>> information to avoid connecting 2 customers with each other (IE the
>>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>>> there...
>>>>
>>>> The closest I can get to implementing the above scenario is by using
>>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>>> receptionist phone to "notify" her that someone is already parked in a
>>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>>> system, and if she fat fingers her transfer, she will double up a queue,
>>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>>> becomes impossible to appropriately handle call park.  If the receptionist
>>>> is away from her desk and attempts to answer a call, she can't tell which
>>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>>> users are prohibited from using call park as they cannot know which slots
>>>> are free (the need for this arises often in certain industries, such as law,
>>>> client is speaking with attorney, attorney finishes and says "let me connect
>>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>>> you.", paralegal is not at desk, atty needs to park call and page his
>>>> assistant, maybe they are in the copy room, or the library doing
>>>> research...)
>>>>
>>>> The asterisk parking lot seems to solve this problem quite elegantly
>>>> (although the stability issues of asterisk undermine it).  It is simple to
>>>> use (receptionist and everyone else only needs to remember 1 extension to
>>>> park any call), The callers/parking spots are maintained by the system to
>>>> prevent double parking, and each caller sits at a specific extension until
>>>> they are picked up or time out.
>>>>
>>>> So, the question boils down to how do people using freeswitch view/use
>>>> call park?  Is there a "better" roaming feature that I just don't know
>>>> about?  If it is just something that hasn't been implemented/thought about
>>>> this way yet, any idea what it would take to get a call parking feature that
>>>> satisfies these requirements into freeswitch?
>>>>
>>>> Thanks for your time, sorry for the long email, hopefully it explains
>>>> well enough what I'm trying to accomplish.
>>>>
>>>> -Tom
>>>>
>>>
>>> Tom,
>>>
>>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>>> Secondly, thank you for actually looking at the documentation before asking
>>> your question.
>>>
>>> You've got yourself quite the little conundrum. Many of us use park in a
>>> more limited role, or we rely on the receptionist not to do silly things
>>> like sending one external caller to the slot of another external caller. We
>>> all agree that's a bad idea, unless it is done deliberately to connect two
>>> external parties. It sounds to me like this particular situation needs an
>>> operator panel app where the receptionist can have a pretty web page showing
>>> what calls are parked where. To my knowledge, this does not yet exist in
>>> FreeSWITCH. The good news is that FS provides all the requisite building
>>> blocks to create such an application using the event socket. How do you feel
>>> about doing a little development? :)
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Mar 17 08:46:49 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 10:46:49 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
Message-ID: <85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>

Call start answer time minus call start time = ring time in the CDR

/b

On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:

> I'm converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I've worked out how to do nearly everything, except for tracking one important metric: How long the phone rang before an agent picked it up.




From jmesquita at freeswitch.org  Wed Mar 17 08:50:25 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 12:50:25 -0300
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
Message-ID: <ebbc2d551003170850v2a0ab9fagb099cb9425fe78b3@mail.gmail.com>

Look at the Local_Extension in dialplan/default.xml

The regex doesn't match 8000 so you can call from one to another. All you
gotta do is change that and issue a reloadxml on fs_cli.

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 12:29 PM, false <cucku.cucku at yahoo.com.vn> wrote:

> HI list
>
> i am new to Freeswitch, i am successfully install freeswitch
> and i use the default config
> i use 1000 and 1001 to register successfull and make call between 2
> endpoints fine
>
> after that, i create a new 8000.xml file which in the same location as
> 1000.xml
> freeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml
> <include>
>   <user id="8000">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="vm-password" value="1000"/>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="8000"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 8000"/>
>       <variable name="effective_caller_id_number" value="8000"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
> the 8000 is registered successfull but i cannot use the 1000 call to 8000
>
> please help
> Ha`
>
> B?n c? blog tr?n Yahoo! 360plus ch?a?
> ??ng k?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Wed Mar 17 08:51:43 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 11:51:43 -0400
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
Message-ID: <8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>

Adjust the dial plan to call extension  8000. The default Dial plan like
below

<condition field="destination_number" expression="^(10[01][0-9])$">

change to

<condition field="destination_number" expression="^(*[1-8]*0[01][0-9])$">

then reloadxml

Hope this help you.

Lloyd

On Wed, Mar 17, 2010 at 11:29 AM, false <cucku.cucku at yahoo.com.vn> wrote:

> HI list
>
> i am new to Freeswitch, i am successfully install freeswitch
> and i use the default config
> i use 1000 and 1001 to register successfull and make call between 2
> endpoints fine
>
> after that, i create a new 8000.xml file which in the same location as
> 1000.xml
> freeswitch:/usr/local/freeswitch/conf/directory# cat default/8000.xml
> <include>
>   <user id="8000">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="vm-password" value="1000"/>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="domestic,international,local"/>
>       <variable name="accountcode" value="8000"/>
>       <variable name="user_context" value="default"/>
>       <variable name="effective_caller_id_name" value="Extension 8000"/>
>       <variable name="effective_caller_id_number" value="8000"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>       <variable name="callgroup" value="techsupport"/>
>     </variables>
>   </user>
> </include>
>
> the 8000 is registered successfull but i cannot use the 1000 call to 8000
>
> please help
> Ha`
>
> B?n c? blog tr?n Yahoo! 360plus ch?a?
> ??ng k?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at fowler.cc  Wed Mar 17 08:53:09 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 17 Mar 2010 11:53:09 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>

I was thrilled when mod_silk was released... but my joy was short lived when I got this reply from Skype about SILK support on their "Skype for SIP beta".


"We have double checked this with Skype for SIP developers.  They said that at the moment SILK codecs are not yet supported, even though we have information about this on the web site.



Most probably SILK codecs will be supported when Skype for SIP goes out from beta stage."



Just wanted to share to avoid frustration for other FreeSWITCHers.

Chris.

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From brian at freeswitch.org  Wed Mar 17 08:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 10:56:56 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
Message-ID: <8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>

Thats funny their own internal team doesn't implement the codec as quickly as we did?  har har har

/b

On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:

> Just wanted to share to avoid frustration for other FreeSWITCHers.

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From brian at freeswitch.org  Wed Mar 17 09:20:10 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 11:20:10 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
Message-ID: <18E6B133-7DF4-4079-9D33-1004FF501379@freeswitch.org>

I just updated our code and module to the official release source from Skype.  They did make a few modifications.

/b

On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:

> I was thrilled when mod_silk was released? but my joy was short lived when I got this reply from Skype about SILK support on their ?Skype for SIP beta?.
>  
> ?We have double checked this with Skype for SIP developers.  They said that at the moment SILK codecs are not yet supported, even though we have information about this on the web site.
>  
> Most probably SILK codecs will be supported when Skype for SIP goes out from beta stage.?
>  
>  
> Just wanted to share to avoid frustration for other FreeSWITCHers.
>  
> Chris.

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From infos at madovsky.org  Wed Mar 17 09:29:56 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 12:29:56 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
	<8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>
Message-ID: <3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>

I think when you have millions of users they need more attention before to release anything.
I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
so it's better to be sure that the bug is well corrected...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 11:56 AM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Thats funny their own internal team doesn't implement the codec as quickly as we did?  har har har


  /b


  On Mar 17, 2010, at 10:53 AM, Chris Fowler wrote:


    Just wanted to share to avoid frustration for other FreeSWITCHers.




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Wed Mar 17 09:33:59 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 11:33:59 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net>
	<8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org>
	<3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
Message-ID: <E367901D-1D54-4BB8-80A3-3EB86D79333B@freeswitch.org>

Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P

/b

On Mar 17, 2010, at 11:29 AM, Madovsky wrote:

> I think when you have millions of users they need more attention before to release anything.
> I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
> so it's better to be sure that the bug is well corrected...

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From infos at madovsky.org  Wed Mar 17 09:44:11 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 12:44:11 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <7454A296C7EDE34EA57199FAA401E2F11C640C7907@VMBX113.ihostexchange.net><8FD55570-8854-4042-9F40-E83E16983AE7@freeswitch.org><3B9ED77A4E774131886EAAB0CC71E8BB@MOBILEE1705>
	<E367901D-1D54-4BB8-80A3-3EB86D79333B@freeswitch.org>
Message-ID: <856BB38F378B4E5DBD047D4AC90D9E62@MOBILEE1705>

I wish also for you guys...
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 12:33 PM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P


  /b


  On Mar 17, 2010, at 11:29 AM, Madovsky wrote:


    I think when you have millions of users they need more attention before to release anything.
    I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
    so it's better to be sure that the bug is well corrected...




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Mar 17 09:57:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 11:57:00 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
Message-ID: <191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>

even better you have the progress and progress_media timestamps too
so you can measure from the instance you got the first ringing indication
so then you can also measure how long it took to start ringing too (PDD)


On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org> wrote:

> Call start answer time minus call start time = ring time in the CDR
>
> /b
>
> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>
> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
> and javascript dialplans) and I think I've worked out how to do nearly
> everything, except for tracking one important metric: How long the phone
> rang before an agent picked it up.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Wed Mar 17 10:01:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 12:01:46 -0500
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
	<2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
Message-ID: <191c3a031003171001l3b2f5911v6e47402419781819@mail.gmail.com>

We don't really care what he does with the code.
But if he expects us to help him with a forked tree, it's not going to
happen.

We expect people who make any meaningful changes to the FS core to be wise
enough to work with
us to keep the code in tree so the community can continue to maintain it and
so it can grow with the rest of the
project.

The GPL license is a whole different topic I will avoid at the moment.


On Wed, Mar 17, 2010 at 10:05 AM, Madovsky <infos at madovsky.org> wrote:

> So why not GPL or Gnu Affero public licence ?
>
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, March 17, 2010 9:23 AM
> Subject: Re: [Freeswitch-users] freeswitch to cisco gateway
>
>
> > This is where it gets hairy... Yes the MPL says you have to share your
> > modifications with the original author but as far as I can tell you don't
> > have to distribute to cause this clause to go into effect.
> >
> > Its like this:
> >
> > You change switch_ivr_originate.c, you HAVE to share those changes with
> > Anthony.
> > You create your own mod_kewlshit.c, its yours to do with what you want
> and
> > you don't have to share.
> >
> > You change any code that is included with the download YOU MUST share
> with
> > the original author.
> >
> > /b
> >
> >
> > On Mar 16, 2010, at 11:30 PM, jay binks wrote:
> >
> >> Im not on the FS dev team, but I think you may be a little mistaken.
> >>
> >> generally your free to make modifications to open source code ( for
> >> internal use )
> >> however its when you distribute the resultant product, then you have
> >> to make the source available..
> >>
> >> if your only using it in-house, I dont believe your required to
> >> contribute the code back.
> >> ( however your going to end up having issues maintaining your patch
> >> set over a long period.. like can be seen here )
> >>
> >> see here :
> >>
> >>> must be made available in Source Code form under the terms of this
> >>> License either on the
> >>> same media as an Executable version
> >>
> >> if your not distributing your executable to any 3rd parties... then I
> >> dont believe you have to distribute the modified source..
> >>
> >> Correct me if Im wrong though guys...
> >>
> >> Jay
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Wed Mar 17 10:11:36 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 13:11:36 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
Message-ID: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>

But Brian,

are you ready to answer to 40 millions users if  the bug is not resolved ? :D
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 12:44 PM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  I wish also for you guys...
    ----- Original Message ----- 
    From: Brian West 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 17, 2010 12:33 PM
    Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


    Wish our FreeSWITCH users would do the same.  Seems we are close to a release and a flood of bugs come in.  :P 


    /b


    On Mar 17, 2010, at 11:29 AM, Madovsky wrote:


      I think when you have millions of users they need more attention before to release anything.
      I know a guy who works for google chrome project he said every bug there are 40 millions people who are waiting to test new version...
      so it's better to be sure that the bug is well corrected...




----------------------------------------------------------------------------


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org
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From brian at freeswitch.org  Wed Mar 17 10:15:52 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 12:15:52 -0500
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
Message-ID: <6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>

Sure... I would love that.

/b

On Mar 17, 2010, at 12:11 PM, Madovsky wrote:

> But Brian,
>  
> are you ready to answer to 40 millions users if  the bug is not resolved ? :D
> 

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From maciej.aniserowicz at gmail.com  Wed Mar 17 10:17:46 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 17 Mar 2010 09:17:46 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
Message-ID: <1268846266197-4751694.post@n2.nabble.com>


Hi Brian,
No, it's not trunk, I updated ~2 weeks ago.

Is this a known issue and updating to trunk will fix this problem?

I tried Anthony's suggestion with waiting for park AND answer events to
begin call processing... and I pasted another scenario. Call are bridged
correcly and can talk to each other, but hanging up 2nd call ends up in
hanging up the first one with ORIGINATOR_CANCEL cause...
Here it is, < 290 lines of commented logs:
http://pastebin.freeswitch.org/12468


-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751694.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From fraserredmond at gmail.com  Wed Mar 17 10:17:54 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 17:17:54 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
Message-ID: <c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>

Yeah, that's what I had expected, and was why I was confused.

I've done some more tests, and tracked the results a little closer in a
spreadsheet, and it seems that when I call the extension directly from
another sipphone the  Caller-Channel-Answered-Time minus
Caller-Channel-Created-Time  matches the ring time, so that scenario is
fine.

When I call into an IVR with the javascript dialplan and then create the new
session and bridge them that way, the Answered-Time and Created-Time, that
are reported after the call ends are reported on the A-leg's Created &
Answered.

The good news is that the Progress-Time is reported on when the B-leg
started ringing, so I can know when the call started ringing.
The bad news is that the Progress-Media-Time is always blank

I can take the Progress-Time and compare it to the system-clock, and that
should generally be accurate to a second or so, which is better than
nothing, but it'd be nice to do it properly.

Any other ideas? Is there any channel-variables I can set that would be
worth playing with? I'm currently only setting:
ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true

Cheers,
Fraser




On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> even better you have the progress and progress_media timestamps too
> so you can measure from the instance you got the first ringing indication
> so then you can also measure how long it took to start ringing too (PDD)
>
>
> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Call start answer time minus call start time = ring time in the CDR
>>
>> /b
>>
>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>
>> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
>> and javascript dialplans) and I think I've worked out how to do nearly
>> everything, except for tracking one important metric: How long the phone
>> rang before an agent picked it up.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Mar 17 10:38:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 12:38:34 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268846266197-4751694.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
Message-ID: <3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>

Why don't you update and see... this problem sounds familiar but without you updating I can't 100% recall.

When we say update... be like Nike and JUST DO IT!  ;)

/b

On Mar 17, 2010, at 12:17 PM, Maciej Aniserowicz wrote:

> 
> Hi Brian,
> No, it's not trunk, I updated ~2 weeks ago.
> 
> Is this a known issue and updating to trunk will fix this problem?
> 
> I tried Anthony's suggestion with waiting for park AND answer events to
> begin call processing... and I pasted another scenario. Call are bridged
> correcly and can talk to each other, but hanging up 2nd call ends up in
> hanging up the first one with ORIGINATOR_CANCEL cause...
> Here it is, < 290 lines of commented logs:
> http://pastebin.freeswitch.org/12468

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From msc at freeswitch.org  Wed Mar 17 10:50:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 10:50:49 -0700
Subject: [Freeswitch-users] how to access session variables after
	hangup???
In-Reply-To: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
References: <50e456911003141301l1003a2a4n5ad00fd4f161b774@mail.gmail.com>
Message-ID: <87f2f3b91003171050h5d791ab7t409d39c43d1abdf2@mail.gmail.com>

On Sun, Mar 14, 2010 at 1:01 PM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> > Pass them as arguments to your luarun hangup hook.
> And what about variables are not available before the call setup (like
> uuid), only during the call progress or after the hangup???
> I would need to access the originate_disposition variable...
>
> >> 'luarun hangup.lua ${originate_disposition}'
> >> 2010-03-14 18:25:08.122511 [CRIT] switch_channel.c:759 Invalid data
> (${api_hangup_hook} contains a variable)
> any other hint???
>

Use the magic env object, which is an event object with all the chan vars
for the channel that just hung up. More info here:
http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object
-MC
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From maciej.aniserowicz at gmail.com  Wed Mar 17 10:59:00 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 17 Mar 2010 09:59:00 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
Message-ID: <1268848740857-4751927.post@n2.nabble.com>


OK, call me Nike;). I'll do that in the upcoming days and see if the problem
stays.

The reason I'm not downloading trunk very frequently is that almost every
time I update, something just stops working. So once I am in a
"pretty-stable" state I don't think about FS update for some time.
And, well, this last update wasn't different from the previous ones as you
can see :).

I'll post a message in this subject to let you know if it works with the
latest bits.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751927.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Wed Mar 17 11:11:08 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 13:11:08 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268848740857-4751927.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
Message-ID: <90C5E5AA-C6A5-48BF-9C73-C1807676D9DE@freeswitch.org>


On Mar 17, 2010, at 12:59 PM, Maciej Aniserowicz wrote:

> 
> OK, call me Nike;). I'll do that in the upcoming days and see if the problem
> stays.
> 
> The reason I'm not downloading trunk very frequently is that almost every
> time I update, something just stops working. So once I am in a
> "pretty-stable" state I don't think about FS update for some time.

I suspect you're doing a dirty update and have build skew.  How exactly are you doing your updates?  Trunk has been very stable and only getting better moving forward... if you had real issues you should be reporting them in jira.  Not ignoring the issues.  Thats how things don't get fixed.  I would rather you open a jira and let me see the issue and close it with "Not A Bug" then chance letting a bug slip thru.

> And, well, this last update wasn't different from the previous ones as you
> can see :).
> 
> I'll post a message in this subject to let you know if it works with the
> latest bits.
> -- 




From anthony.minessale at gmail.com  Wed Mar 17 11:31:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:31:39 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268848740857-4751927.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
Message-ID: <191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>

What If I used that same policy that you use to update when it came to
answering your emails?

Please update to latest trunk so I can confirm your traces against the
latest code.
Also run the command show channels while it's bridged.

I would prefer if you do it ASAP, now that you have us worrying about your
issue, its your duty to see it through.




On Wed, Mar 17, 2010 at 12:59 PM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> OK, call me Nike;). I'll do that in the upcoming days and see if the
> problem
> stays.
>
> The reason I'm not downloading trunk very frequently is that almost every
> time I update, something just stops working. So once I am in a
> "pretty-stable" state I don't think about FS update for some time.
> And, well, this last update wasn't different from the previous ones as you
> can see :).
>
> I'll post a message in this subject to let you know if it works with the
> latest bits.
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4751927.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Mar 17 11:47:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:47:35 -0500
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
Message-ID: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>

side note:

Some of mod_valet_parking seems undocumented you can do the following

 <extension name="park-in">
      <condition field="destination_number" expression="^(8000)$">
        <action application="valet_park" data="auto in 8001 8999"/>
      </condition>
    </extension>


 <extension name="park-out">
     <condition field="destination_number" expression="^(8\d{3})$">
        <action application="answer"/>
        <action application="valet_park" data="test $1"/>
     </condition>
</extension>


So if you attended transfer to 8000, you will park them at the next
available slot
This is missing the blf like fifo has but it will be unique announced slots.


also

you can add this before any call

 <action application="bind_meta_app" data="1 a s valet_park::test auto in
8001 8999"/>

so when on the phone you can dial *1 to park the call to the next free slot
8001-8999




On Tue, Mar 16, 2010 at 9:17 PM, Tom Christensen <pavera at gmail.com> wrote:

>
>
> On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>
>>> Hello,
>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>> is really nice, straightforward, and I appreciate the design of the system.
>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>> really looking for something more stable, and I think I've found it here!
>>>
>>> One issue I've run into, and I'm not sure if its just that my use case is
>>> abnormal, hence the email.
>>>
>>> I've worked with PBX systems for a number of years (11 now...), and in
>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>> feature.  By this I mean, you think party X (the intended recipient of the
>>> call) is in the building, but they are not at their desk.  So, you park the
>>> call, and then page, or by some other means attempt to locate party X and
>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>> party X or you can't find them in time, the call should time out back to the
>>> parking party (receptionist) to be handled appropriately (message taken,
>>> sent to voicemail, transferred to party X's boss)
>>>
>>> This scenario requires a few things:
>>> 1) the receptionist/person who actually receives the call needs to be
>>> able to put the call in a specific place (extension) that doesn't change
>>> 2) Party X (the intended recipient) needs to be able to access that
>>> specific place (extension) from any phone inside the company
>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>> party X and one for party Y, in that order, but party Y gets to a phone
>>> first, they will be connected to party X's caller)
>>> 4) The receptionist has to be protected from or provided with sufficient
>>> information to avoid connecting 2 customers with each other (IE the
>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>> there...
>>>
>>> The closest I can get to implementing the above scenario is by using
>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>> receptionist phone to "notify" her that someone is already parked in a
>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>> system, and if she fat fingers her transfer, she will double up a queue,
>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>> becomes impossible to appropriately handle call park.  If the receptionist
>>> is away from her desk and attempts to answer a call, she can't tell which
>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>> users are prohibited from using call park as they cannot know which slots
>>> are free (the need for this arises often in certain industries, such as law,
>>> client is speaking with attorney, attorney finishes and says "let me connect
>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>> you.", paralegal is not at desk, atty needs to park call and page his
>>> assistant, maybe they are in the copy room, or the library doing
>>> research...)
>>>
>>> The asterisk parking lot seems to solve this problem quite elegantly
>>> (although the stability issues of asterisk undermine it).  It is simple to
>>> use (receptionist and everyone else only needs to remember 1 extension to
>>> park any call), The callers/parking spots are maintained by the system to
>>> prevent double parking, and each caller sits at a specific extension until
>>> they are picked up or time out.
>>>
>>> So, the question boils down to how do people using freeswitch view/use
>>> call park?  Is there a "better" roaming feature that I just don't know
>>> about?  If it is just something that hasn't been implemented/thought about
>>> this way yet, any idea what it would take to get a call parking feature that
>>> satisfies these requirements into freeswitch?
>>>
>>> Thanks for your time, sorry for the long email, hopefully it explains
>>> well enough what I'm trying to accomplish.
>>>
>>> -Tom
>>>
>>
>> Tom,
>>
>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>> Secondly, thank you for actually looking at the documentation before asking
>> your question.
>>
>> You've got yourself quite the little conundrum. Many of us use park in a
>> more limited role, or we rely on the receptionist not to do silly things
>> like sending one external caller to the slot of another external caller. We
>> all agree that's a bad idea, unless it is done deliberately to connect two
>> external parties. It sounds to me like this particular situation needs an
>> operator panel app where the receptionist can have a pretty web page showing
>> what calls are parked where. To my knowledge, this does not yet exist in
>> FreeSWITCH. The good news is that FS provides all the requisite building
>> blocks to create such an application using the event socket. How do you feel
>> about doing a little development? :)
>>
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Michael,
> Thanks for the reply. I worked on it more with the business in question
> today and we came up with a hybrid solution which I think will cover all the
> bases.
> 1) I implemented a "call park" extension on each phone, these are standard
> mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
> freeswitch config here BTW, like 8 lines in the dialplan handles this for
> all 50 extensions).  This way everyone has a call park extension that can be
> picked up anywhere, the numbering scheme is essentially take your extension
> prepend a 2, that is your call park extension, so if someone pages you, you
> just pick up any phone and dial 2+ext and that will pick the call up.
>  Everyone in the office can use this feature to park a call for anyone,
> because only calls for a specific person will ever be parked in that queue,
> you can put multiple people there without issue.
> 2) Implemented as mentioned a "parking lot" set of mod_fifo queues that
> have SLA/BLA set up on the receptionist phone.  She can use these queues to
> quickly and easily park a bunch of people at once (IE when there are call
> floods) and then send them off to where they need to go as time permits.  If
> she isn't at her phone, she can still answer calls and park them onto
> individual users park extensions.  Still doesn't have the "safety" of not
> double parking someone in the parking lot, but we'll trust her to be able to
> see red lights :)
>
> This whole thing goes live Thursday, so wish me luck :)
> Thanks again!
> Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anthony.minessale at gmail.com  Wed Mar 17 11:52:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 13:52:23 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
Message-ID: <191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>

progress_media_time is only set if you get a 183
if you get a 180 with no media it uses progress_time instead

as long as you have one or the other you have the timestamp of when it
started ringing.





On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Yeah, that's what I had expected, and was why I was confused.
>
> I've done some more tests, and tracked the results a little closer in a
> spreadsheet, and it seems that when I call the extension directly from
> another sipphone the  Caller-Channel-Answered-Time minus
> Caller-Channel-Created-Time  matches the ring time, so that scenario is
> fine.
>
> When I call into an IVR with the javascript dialplan and then create the
> new session and bridge them that way, the Answered-Time and Created-Time,
> that are reported after the call ends are reported on the A-leg's Created &
> Answered.
>
> The good news is that the Progress-Time is reported on when the B-leg
> started ringing, so I can know when the call started ringing.
> The bad news is that the Progress-Media-Time is always blank
>
> I can take the Progress-Time and compare it to the system-clock, and that
> should generally be accurate to a second or so, which is better than
> nothing, but it'd be nice to do it properly.
>
> Any other ideas? Is there any channel-variables I can set that would be
> worth playing with? I'm currently only setting:
> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> even better you have the progress and progress_media timestamps too
>> so you can measure from the instance you got the first ringing indication
>> so then you can also measure how long it took to start ringing too (PDD)
>>
>>
>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Call start answer time minus call start time = ring time in the CDR
>>>
>>> /b
>>>
>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>
>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using xml
>>> and javascript dialplans) and I think I've worked out how to do nearly
>>> everything, except for tracking one important metric: How long the phone
>>> rang before an agent picked it up.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From lloyd.aloysius at gmail.com  Wed Mar 17 11:54:41 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 14:54:41 -0400
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
Message-ID: <8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>

 <action application="bridge" data="user/${dialed_extension}@
${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
mber}"/>

I setup the above dial plan. The call to extension works perfect. But It
never call to the ${followme_neumber}.

How can I fix this issue.

Thanks
Lloyd




On Mon, Mar 8, 2010 at 7:24 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Michael,
>
> Thank you for the suggestion. This suggestion make sense and a proper way
> to implement the follow me. .
>
> Thanks
> Lloyd
>
> On Mon, Mar 8, 2010 at 6:10 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Mar 8, 2010 at 2:33 PM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>>
>>> Hi All,
>>>
>>> I would like to setup follow me dialplan for individual extensions. Some
>>> extensions need a follow-me and some not. If the the follow-me not reach any
>>> live person the call should go to the FreeSWTICH voicemail.I have the
>>> extensions range from 201 -209. Also I am using the following Default dial
>>> plan to call between extensions.
>>>
>>> <extension name="Local_Extension">
>>>       <condition field="destination_number" expression="^(20[0-9])$">
>>> <action application="set" data="dialed_extension=$1"/>
>>>  <action application="export" data="dialed_extension=$1"/>
>>> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
>>>  <action application="bind_meta_app" data="1 b s execute_extension::dx
>>> XML features"/>
>>> <action application="bind_meta_app" data="2 b s
>>> record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>>  <action application="bind_meta_app" data="3 b s execute_extension::cf
>>> XML features"/>
>>> <action application="set" data="ringback=${us-ring}"/>
>>>  <action application="set" data="transfer_ringback=$${hold_music}"/>
>>> <action application="set" data="call_timeout=30"/>
>>>  <!-- <action application="set"
>>> data="sip_exclude_contact=${network_addr}"/> -->
>>> <action application="set" data="hangup_after_bridge=true"/>
>>>  <!--<action application="set"
>>> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
>>> -->
>>>  <action application="set" data="continue_on_fail=true"/>
>>> <action application="hash"
>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>  <action application="hash"
>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>> <action application="set"
>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>> var callgroup)}"/>
>>>  <!--<action application="export"
>>> data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name}
>>> var sip_secure_media)}"/>-->
>>>  <action application="hash"
>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>> <action application="bridge" data="user/${dialed_extension}@
>>> ${domain_name},${followme}"/>
>>>  <action application="answer"/>
>>> <action application="sleep" data="1000"/>
>>>  <action application="voicemail" data="default ${domain_name}
>>> ${dialed_extension}"/>
>>>       </condition>
>>>  </extension>
>>>
>>> I use the wiki samples to make a follow me dial plan for extension 201.
>>> Here is the dial plan.
>>>
>>> <extension name="201_followme">
>>>   <condition field="destination_number" expression="201">
>>>    <action application="set" data="hangup_after_bridge=true"/>
>>>    <action application="set" data="continue_on_fail=true"/>
>>>    <action application="set" data="ignore_early_media=true"/>
>>>    <action application="set" data="call_timeout=10"/>
>>>    <action application="bridge" data="sofia/$${domain}/201"/>
>>>    <action application="set" data="call_timeout=13"/>
>>>    <action application="bridge"
>>> data="sofia/gateway/voicepulse/14161235555" />
>>>    <action application="answer"/>
>>>    <action application="sleep" data="1000"/>
>>>    <action application="voicemail" data="default $${domain} 201"/>
>>>   </condition>
>>> </extension>
>>>
>>>
>>> Call to extension 201 , there are two dialplan matching the request.
>>> Either Local_Extension or follwme_201. Which order freeswtich select?
>>>
>>
>> It will match whichever one comes first in the file. You could, for
>> example, put this 201 extension above the Local_Extension. You could also
>> put a custom chan var in the user directory for those who need the
>> follow-me. Then you could have two conditions, both of which need to match:
>>
>> <extension name="Local, follow me">
>>   <condition field="${followme}" expression="^yes$"/>
>>   <condition field="destination_number expression="^(20\d)$">
>>     <!-- followme stuff here -->
>>     ...
>>   </condition>
>> </extension>
>>
>> <extension name="Local No follow me">
>>   <condition field="${followme}" expression="^$"/>
>>   <condition field="destination_number expression="^(20\d)$">
>>     <!-- normal local extension stuff here -->
>>     ...
>>   </condition>
>> </extension>
>>
>> Then in the user setup in the directory add the followme variable only to
>> those who need it. You can also add a variable for the followme phone number
>> if you want. Inside the <variables> node just add whatever you want:
>>
>>   <variable name="followme" value="yes"/>
>>   <variable name="followme_number" value="14161235555"/>
>> Or even
>>   <variable name="followme_dialstring"
>> value="sofia/gateway/voicepulse/14161235555"/>
>>
>> Then you could use ${followme_number} in your bridge like this:
>>
>>   <action application="bridge" data="sofia/gateway/
>> voicepulse/${followme_number}" />
>>
>> Or ${followme_dialstring} like this:
>>   <action application="bridge" data="${followme_dialstring}" />
>>
>> Make sense? :)
>>
>> Give it a try and report back your success/failure/questions/comments/etc.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From lloyd.aloysius at gmail.com  Wed Mar 17 12:07:05 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 17 Mar 2010 15:07:05 -0400
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
	<191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
Message-ID: <8a19bf2e1003171207q2432918g90830c33afd5bdcc@mail.gmail.com>

Thanks Antony.

Lloyd


On Wed, Mar 17, 2010 at 2:47 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> side note:
>
> Some of mod_valet_parking seems undocumented you can do the following
>
>  <extension name="park-in">
>       <condition field="destination_number" expression="^(8000)$">
>         <action application="valet_park" data="auto in 8001 8999"/>
>       </condition>
>     </extension>
>
>
>  <extension name="park-out">
>      <condition field="destination_number" expression="^(8\d{3})$">
>         <action application="answer"/>
>         <action application="valet_park" data="test $1"/>
>      </condition>
> </extension>
>
>
> So if you attended transfer to 8000, you will park them at the next
> available slot
> This is missing the blf like fifo has but it will be unique announced
> slots.
>
>
> also
>
> you can add this before any call
>
>  <action application="bind_meta_app" data="1 a s valet_park::test auto in
> 8001 8999"/>
>
> so when on the phone you can dial *1 to park the call to the next free slot
> 8001-8999
>
>
>
>
>
> On Tue, Mar 16, 2010 at 9:17 PM, Tom Christensen <pavera at gmail.com> wrote:
>
>>
>>
>> On Tue, Mar 16, 2010 at 3:14 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Mon, Mar 15, 2010 at 11:59 PM, Tom Christensen <pavera at gmail.com>wrote:
>>>
>>>> Hello,
>>>> Pretty new, but so far very happy with freeswitch!  The setup and config
>>>> is really nice, straightforward, and I appreciate the design of the system.
>>>>  I've worked with Asterisk pretty regularly for the last 4-5 years, and am
>>>> really looking for something more stable, and I think I've found it here!
>>>>
>>>> One issue I've run into, and I'm not sure if its just that my use case
>>>> is abnormal, hence the email.
>>>>
>>>> I've worked with PBX systems for a number of years (11 now...), and in
>>>> all of the systems I've worked with, call parking was kind of a "roaming"
>>>> feature.  By this I mean, you think party X (the intended recipient of the
>>>> call) is in the building, but they are not at their desk.  So, you park the
>>>> call, and then page, or by some other means attempt to locate party X and
>>>> tell them "get to the nearest phone and pick up 5901".  If you can't locate
>>>> party X or you can't find them in time, the call should time out back to the
>>>> parking party (receptionist) to be handled appropriately (message taken,
>>>> sent to voicemail, transferred to party X's boss)
>>>>
>>>> This scenario requires a few things:
>>>> 1) the receptionist/person who actually receives the call needs to be
>>>> able to put the call in a specific place (extension) that doesn't change
>>>> 2) Party X (the intended recipient) needs to be able to access that
>>>> specific place (extension) from any phone inside the company
>>>> 3) Order must be preserved (IE, the mod_fifo call park in freeswitch is
>>>> inadequate, because if the receptionist parks 2 people on 5900, one for
>>>> party X and one for party Y, in that order, but party Y gets to a phone
>>>> first, they will be connected to party X's caller)
>>>> 4) The receptionist has to be protected from or provided with sufficient
>>>> information to avoid connecting 2 customers with each other (IE the
>>>> valet_park in freeswitch is inadequate, because if she parks a caller for X
>>>> on 6001, then parks a caller for Y on 6002, then a caller for Z calls in,
>>>> how can she know if 6001 is free yet? valet_park doesn't support BLA/SLA so
>>>> what can she do?  What if she just forgets and sends that caller to 6001?
>>>>  Then the 2 customers are bridged together and thats a big no-no)  Also, I
>>>> haven't seen a way to make valet_park timeout?  But maybe I'm wrong
>>>> there...
>>>>
>>>> The closest I can get to implementing the above scenario is by using
>>>> multiple mod_fifo queues (one for each parking spot), and SLA/BLA on the
>>>> receptionist phone to "notify" her that someone is already parked in a
>>>> certain queue.  Unfortunately, there is no enforcement of this policy by the
>>>> system, and if she fat fingers her transfer, she will double up a queue,
>>>> creating the problem indicated in 3 above.  The other shortcoming of this
>>>> setup is that without a large phone (IE, with 10-15 SLA/BLA slots) it
>>>> becomes impossible to appropriately handle call park.  If the receptionist
>>>> is away from her desk and attempts to answer a call, she can't tell which
>>>> parking spots are free at all on a regular 2 line phone.  Also, regular
>>>> users are prohibited from using call park as they cannot know which slots
>>>> are free (the need for this arises often in certain industries, such as law,
>>>> client is speaking with attorney, attorney finishes and says "let me connect
>>>> you with my assistant to schedule an appointment and handle x, y, and z for
>>>> you.", paralegal is not at desk, atty needs to park call and page his
>>>> assistant, maybe they are in the copy room, or the library doing
>>>> research...)
>>>>
>>>> The asterisk parking lot seems to solve this problem quite elegantly
>>>> (although the stability issues of asterisk undermine it).  It is simple to
>>>> use (receptionist and everyone else only needs to remember 1 extension to
>>>> park any call), The callers/parking spots are maintained by the system to
>>>> prevent double parking, and each caller sits at a specific extension until
>>>> they are picked up or time out.
>>>>
>>>> So, the question boils down to how do people using freeswitch view/use
>>>> call park?  Is there a "better" roaming feature that I just don't know
>>>> about?  If it is just something that hasn't been implemented/thought about
>>>> this way yet, any idea what it would take to get a call parking feature that
>>>> satisfies these requirements into freeswitch?
>>>>
>>>> Thanks for your time, sorry for the long email, hopefully it explains
>>>> well enough what I'm trying to accomplish.
>>>>
>>>> -Tom
>>>>
>>>
>>> Tom,
>>>
>>> First of all, welcome to FreeSWITCH - the future of telephony! :P
>>> Secondly, thank you for actually looking at the documentation before asking
>>> your question.
>>>
>>> You've got yourself quite the little conundrum. Many of us use park in a
>>> more limited role, or we rely on the receptionist not to do silly things
>>> like sending one external caller to the slot of another external caller. We
>>> all agree that's a bad idea, unless it is done deliberately to connect two
>>> external parties. It sounds to me like this particular situation needs an
>>> operator panel app where the receptionist can have a pretty web page showing
>>> what calls are parked where. To my knowledge, this does not yet exist in
>>> FreeSWITCH. The good news is that FS provides all the requisite building
>>> blocks to create such an application using the event socket. How do you feel
>>> about doing a little development? :)
>>>
>>> -MC
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> Michael,
>> Thanks for the reply. I worked on it more with the business in question
>> today and we came up with a hybrid solution which I think will cover all the
>> bases.
>> 1) I implemented a "call park" extension on each phone, these are standard
>> mod_fifo queues with SLA/BLA on each phone (LOVED the power of the
>> freeswitch config here BTW, like 8 lines in the dialplan handles this for
>> all 50 extensions).  This way everyone has a call park extension that can be
>> picked up anywhere, the numbering scheme is essentially take your extension
>> prepend a 2, that is your call park extension, so if someone pages you, you
>> just pick up any phone and dial 2+ext and that will pick the call up.
>>  Everyone in the office can use this feature to park a call for anyone,
>> because only calls for a specific person will ever be parked in that queue,
>> you can put multiple people there without issue.
>> 2) Implemented as mentioned a "parking lot" set of mod_fifo queues that
>> have SLA/BLA set up on the receptionist phone.  She can use these queues to
>> quickly and easily park a bunch of people at once (IE when there are call
>> floods) and then send them off to where they need to go as time permits.  If
>> she isn't at her phone, she can still answer calls and park them onto
>> individual users park extensions.  Still doesn't have the "safety" of not
>> double parking someone in the parking lot, but we'll trust her to be able to
>> see red lights :)
>>
>> This whole thing goes live Thursday, so wish me luck :)
>> Thanks again!
>> Tom
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 12:11:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:11:10 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
Message-ID: <87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>

On Wed, Mar 17, 2010 at 5:22 AM, Vladimir Elizarov
<xengelpublicx at gmail.com>wrote:

> Yes
>
>
You can't. The FIFO system isn't transferring the call to the agent's phone.
Rather, it is ringing the agent's phone and when the agent picks up, it then
bridges the caller to the agent.

What exactly are you trying to accomplish? Just curious. By the way you can
possibly resolve this issue by create an "agent access" extension. The idea
is simple: User A picks up and dials an extension; that extension then
connects User A to the longest waiting caller in the FIFO queue.

Use this example to create your "agent access" extension:
http://wiki.freeswitch.org/wiki/Mod_fifo#Take_a_caller_out_of_a_FIFO_queue

The net result is probably the same: a third phone connects to the caller in
queue. If that caller is the only one in the queue then the agent's phone
that is ringing will stop ringing immediately.

-MC
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From msc at freeswitch.org  Wed Mar 17 12:13:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:13:13 -0700
Subject: [Freeswitch-users] Lua playAndGetDigits
In-Reply-To: <5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
References: <5c9dcbfb1003161321q494ab3a0o5f90278281332e06@mail.gmail.com>
	<5c9dcbfb1003161723m1bcd39d7tf29738aa8b96f19a@mail.gmail.com>
	<8a19bf2e1003161737r39c9fe03k11192ab4aa8f86a9@mail.gmail.com>
	<5c9dcbfb1003161751q4a2e63e5p6320b436180587af@mail.gmail.com>
	<8a19bf2e1003161807m4930cee2o1efdcde244d03c3b@mail.gmail.com>
	<5c9dcbfb1003161816k5c27c0a1jf82599ebc9f02833@mail.gmail.com>
	<8a19bf2e1003161834m475a7fc3x24bdd09259b31095@mail.gmail.com>
	<5c9dcbfb1003161905s7cf0d4edqd2f44504b4cb762@mail.gmail.com>
	<D3F1BEBC-13DF-4707-9D48-43BC2978682B@freeswitch.org>
	<5c9dcbfb1003170700j391b012exb09f66904af7b022@mail.gmail.com>
Message-ID: <87f2f3b91003171213k39cd5431m43f28651f869d943@mail.gmail.com>

On Wed, Mar 17, 2010 at 7:00 AM, Michael De Lorenzo <
delorenzodesign at gmail.com> wrote:

> How do I do that in Lua?  session:execute("dtmf","start") doesn't work.

session:execute("start_dtmf")
-MC
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From fraserredmond at gmail.com  Wed Mar 17 12:18:00 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 19:18:00 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
Message-ID: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>

Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
way?

I've realised since my last email that even having the progress_time doesn't
help without the progress_media_time or some way to find when the ringing
stopped - getting the server timestamp wouldn't happen until the call ended,
at which point I can't split the ring time from the talk time.

Surely there can't be something that Asterisk handles well that FreeSwitch
doesn't? ;-)

I'm looking into the CDR now - might have to parse the log files to get the
end time from there :-(

Cheers,
Fraser




On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> progress_media_time is only set if you get a 183
> if you get a 180 with no media it uses progress_time instead
>
> as long as you have one or the other you have the timestamp of when it
> started ringing.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Yeah, that's what I had expected, and was why I was confused.
>>
>> I've done some more tests, and tracked the results a little closer in a
>> spreadsheet, and it seems that when I call the extension directly from
>> another sipphone the  Caller-Channel-Answered-Time minus
>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>> fine.
>>
>> When I call into an IVR with the javascript dialplan and then create the
>> new session and bridge them that way, the Answered-Time and Created-Time,
>> that are reported after the call ends are reported on the A-leg's Created &
>> Answered.
>>
>> The good news is that the Progress-Time is reported on when the B-leg
>> started ringing, so I can know when the call started ringing.
>> The bad news is that the Progress-Media-Time is always blank
>>
>> I can take the Progress-Time and compare it to the system-clock, and that
>> should generally be accurate to a second or so, which is better than
>> nothing, but it'd be nice to do it properly.
>>
>> Any other ideas? Is there any channel-variables I can set that would be
>> worth playing with? I'm currently only setting:
>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> even better you have the progress and progress_media timestamps too
>>> so you can measure from the instance you got the first ringing indication
>>> so then you can also measure how long it took to start ringing too (PDD)
>>>
>>>
>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> Call start answer time minus call start time = ring time in the CDR
>>>>
>>>> /b
>>>>
>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>
>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>> everything, except for tracking one important metric: How long the phone
>>>> rang before an agent picked it up.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed Mar 17 12:41:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 14:41:36 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>

i'll try again

progress_media and progress are both the same for your purposes
you will only have exactly one of the 2 set.  its the exact time it started
ringing.

for our sake, if we have either of them we will take the one that is not
zero and call it the progress time.

so you have the created time, the progress time , the answered time and the
hangup time


you always have created time and hangup time but not always progress and
answered

so talk time is (hangup_time - (answered_time unless its 0)) or if
answered_time is 0 there is no talk time
ring time is:
         (answered time unless its zero otherwise hangup_time) -
created_time


edit conf/autoload_configs/cdr_csv.conf.xml

uncomment:

<param name="debug" value="true"/>

and restart FS


now when you hangup all your calls you will see the values you have to work
with in the cdr

if you are tying to do billing inside some js script you are using, stop
now, and go learn about cdrs.





On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?
>
> I've realised since my last email that even having the progress_time
> doesn't help without the progress_media_time or some way to find when the
> ringing stopped - getting the server timestamp wouldn't happen until the
> call ended, at which point I can't split the ring time from the talk time.
>
> Surely there can't be something that Asterisk handles well that FreeSwitch
> doesn't? ;-)
>
> I'm looking into the CDR now - might have to parse the log files to get the
> end time from there :-(
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> progress_media_time is only set if you get a 183
>> if you get a 180 with no media it uses progress_time instead
>>
>> as long as you have one or the other you have the timestamp of when it
>> started ringing.
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> Yeah, that's what I had expected, and was why I was confused.
>>>
>>> I've done some more tests, and tracked the results a little closer in a
>>> spreadsheet, and it seems that when I call the extension directly from
>>> another sipphone the  Caller-Channel-Answered-Time minus
>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>> fine.
>>>
>>> When I call into an IVR with the javascript dialplan and then create the
>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>> that are reported after the call ends are reported on the A-leg's Created &
>>> Answered.
>>>
>>> The good news is that the Progress-Time is reported on when the B-leg
>>> started ringing, so I can know when the call started ringing.
>>> The bad news is that the Progress-Media-Time is always blank
>>>
>>> I can take the Progress-Time and compare it to the system-clock, and that
>>> should generally be accurate to a second or so, which is better than
>>> nothing, but it'd be nice to do it properly.
>>>
>>> Any other ideas? Is there any channel-variables I can set that would be
>>> worth playing with? I'm currently only setting:
>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> even better you have the progress and progress_media timestamps too
>>>> so you can measure from the instance you got the first ringing
>>>> indication
>>>> so then you can also measure how long it took to start ringing too (PDD)
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>
>>>>> /b
>>>>>
>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>
>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>> everything, except for tracking one important metric: How long the phone
>>>>> rang before an agent picked it up.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Wed Mar 17 12:46:48 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 17 Mar 2010 20:46:48 +0100
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <4468a6771003171246t76a58d7erc6d4c3fd48a938c1@mail.gmail.com>

SIP 180 = your phone rings (Ringing message)
SIP 183 + sdp = early media or inband ringing (Session Progress message)

You have both vars to watch for.

Ognjen

On Wed, Mar 17, 2010 at 8:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?
>
> I've realised since my last email that even having the progress_time
> doesn't help without the progress_media_time or some way to find when the
> ringing stopped - getting the server timestamp wouldn't happen until the
> call ended, at which point I can't split the ring time from the talk time.
>
> Surely there can't be something that Asterisk handles well that FreeSwitch
> doesn't? ;-)
>
> I'm looking into the CDR now - might have to parse the log files to get the
> end time from there :-(
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> progress_media_time is only set if you get a 183
>> if you get a 180 with no media it uses progress_time instead
>>
>> as long as you have one or the other you have the timestamp of when it
>> started ringing.
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> Yeah, that's what I had expected, and was why I was confused.
>>>
>>> I've done some more tests, and tracked the results a little closer in a
>>> spreadsheet, and it seems that when I call the extension directly from
>>> another sipphone the  Caller-Channel-Answered-Time minus
>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>> fine.
>>>
>>> When I call into an IVR with the javascript dialplan and then create the
>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>> that are reported after the call ends are reported on the A-leg's Created &
>>> Answered.
>>>
>>> The good news is that the Progress-Time is reported on when the B-leg
>>> started ringing, so I can know when the call started ringing.
>>> The bad news is that the Progress-Media-Time is always blank
>>>
>>> I can take the Progress-Time and compare it to the system-clock, and that
>>> should generally be accurate to a second or so, which is better than
>>> nothing, but it'd be nice to do it properly.
>>>
>>> Any other ideas? Is there any channel-variables I can set that would be
>>> worth playing with? I'm currently only setting:
>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>
>>> Cheers,
>>> Fraser
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> even better you have the progress and progress_media timestamps too
>>>> so you can measure from the instance you got the first ringing
>>>> indication
>>>> so then you can also measure how long it took to start ringing too (PDD)
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>
>>>>> /b
>>>>>
>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>
>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>> everything, except for tracking one important metric: How long the phone
>>>>> rang before an agent picked it up.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Wed Mar 17 12:46:47 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 15:46:47 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
Message-ID: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>

Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

Regards

Franck
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 16, 2010 4:35 PM
  Subject: Re: [Freeswitch-users] voicemail options


  ok, I tried 50 times to leave a message,
  it worked only one time.
  It seems that sendmail needs to be free to send the email....

    ----- Original Message ----- 
    From: devel at thom.fr.eu.org 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Tuesday, March 16, 2010 3:59 PM
    Subject: Re: [Freeswitch-users] voicemail options


    Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

    I don't know if it's a priviledge problem, as I got the same results running FS with or without -u.

     

    Fran?ois

     

     

    De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
    Envoy? : mardi 16 mars 2010 20:30
    ? : freeswitch-users at lists.freeswitch.org
    Objet : Re: [Freeswitch-users] voicemail options

     

    are you running FS as root or a regular user?
    maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



    On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

    I noticed that sendmail makes a segfault when FS is trying to send email

    and sendmail has some emails locked in folder queue....

     

    Regards

     

    Franck


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org




    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900



----------------------------------------------------------------------------


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org
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From kristian.kielhofner at gmail.com  Wed Mar 17 12:50:57 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Wed, 17 Mar 2010 15:50:57 -0400
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
Message-ID: <4d15ff861003171250g595d0f0bn3d01386b9d603df@mail.gmail.com>

They are referring to SIP 180 and SIP 183.

It's kind of a mess (it is SIP) but both are non-final response codes
to indicate call progress.  Typically speaking, 180 Ringing is
returned without media and tells the receiving device (FreeSWITCH or
your local SIP phone in this case) to activate a local ringback
generator to inform the caller the remote end is progressing.  180
without media is typically used when calling SIP phones.  They know
that they only actually "ring".

183 is typically used with media.  The media contains audio to tell
the caller what is happening.  Typically this audio sounds just like
ringback (depending on region) but it could just as easily be a
ringback tone (Britney Spears or anything else), audio from the remote
end, or just about anything else the caller might need to know (even
an error message).

Most carriers *should* consistently give you 183 with media while the
call progresses, whether the remote end is actually ringing or not
depends.  You have to listen to the audio :).

BTW, as are most things with SIP these are generalities and many not
happen in your environment, most stacks, for example, will handle
180/183 interchangeably - it's the presence of media (SDP) that
matters.

FreeSWITCH has a multitude of options to deal with these scenarios.
First things first - what does your carrier return?

On Wed, Mar 17, 2010 at 3:18 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:
> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com



From robert.dyck at shaw.ca  Wed Mar 17 12:51:55 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 12:51:55 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003170049.06565.robert.dyck@shaw.ca>
	<ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
Message-ID: <201003171251.57460.robert.dyck@shaw.ca>

I can see a process for fscomm. Should I be seeing a process for the core.

When starting fscomm it reports -
Initializing globals... 

Initializing core... 

Everything OK, Entering runtime loop ...

BTW, not Windows but Fedora 11, KDE flavour

Thanks for looking into this
RD


On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> Rob, I believe that the core has failed to load. I have not added any type
> of check on the code level for those cases yet as I believed it would never
> really fail to load. Those cases have been more frequent than what I
> originally expected so I will add that to code today (hopefully later
> tonight).
> 
> Meanwhile, what you can do is check if the ports are open or not on the OS.
> I am not sure how to do that on Windows. Or, maybe we already have logs on
> the log folder?
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > Is there something wrong with my build?
> > 
> > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > [Socket
> > Connection Error]" with fscomm running.
> > 
> > RD
> > 
> > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > With FSComm running you can use fs_cli to try pa list. If u can't list
> > 
> > the
> > 
> > > devices that would be the problem. I'm going to try to dig into this
> > > further on 1 of my desktops w/ pulseaudio.
> > > 
> > > -- W
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From anthony.minessale at gmail.com  Wed Mar 17 12:55:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 14:55:00 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
Message-ID: <191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>

could you get a svn diff of the switch_utils.c so we can  see what the
change was?



On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

>  Finally I succeed to send voicemail with sendmail through a little php
> script (could be Perl also)
> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
> doesn't exist.
>
> I attached my php script and switch_utils.c. Not sure if the FS developers
> will accept my modification
>
> Regards
>
> Franck
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 4:35 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
> ok, I tried 50 times to leave a message,
> it worked only one time.
> It seems that sendmail needs to be free to send the email....
>
>
> ----- Original Message -----
> *From:* devel at thom.fr.eu.org
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 16, 2010 3:59 PM
> *Subject:* Re: [Freeswitch-users] voicemail options
>
>  Are you running debian, I have the same problem on debian x64 (already
> mentioned here
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
> and cannot fix it.
>
> I don?t know if it?s a priviledge problem, as I got the same results
> running FS with or without ?u.
>
>
>
> Fran?ois
>
>
>
>
>
> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
> Minessale
> *Envoy? :* mardi 16 mars 2010 20:30
> *? :* freeswitch-users at lists.freeswitch.org
> *Objet :* Re: [Freeswitch-users] voicemail options
>
>
>
> are you running FS as root or a regular user?
> maybe there is a permission or resource limitation problem preventing it
> from getting the larger stack space necessary?
>
>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>
> I noticed that sendmail makes a segfault when FS is trying to send email
>
> and sendmail has some emails locked in folder queue....
>
>
>
> Regards
>
>
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Wed Mar 17 12:58:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 12:58:45 -0700
Subject: [Freeswitch-users] Call parking question/best practice advice
In-Reply-To: <191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
References: <82dae1c01003152359g5da168cbqf37a6326fe73b785@mail.gmail.com>
	<87f2f3b91003161414q670618e5ia26090784cf9cc94@mail.gmail.com>
	<82dae1c01003161917m3544c94an65931830190e77ba@mail.gmail.com>
	<191c3a031003171147n1b4bec9u8293a9b9c038cabd@mail.gmail.com>
Message-ID: <87f2f3b91003171258x77678688v2515af8cc7b62b29@mail.gmail.com>

FYI,

This method and the bind meta app method are now in the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Valet_Autoselect_Parking_Stall

-MC

On Wed, Mar 17, 2010 at 11:47 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> side note:
>
> Some of mod_valet_parking seems undocumented you can do the following
>
>  <extension name="park-in">
>       <condition field="destination_number" expression="^(8000)$">
>         <action application="valet_park" data="auto in 8001 8999"/>
>       </condition>
>     </extension>
>
>
>  <extension name="park-out">
>      <condition field="destination_number" expression="^(8\d{3})$">
>         <action application="answer"/>
>         <action application="valet_park" data="test $1"/>
>      </condition>
> </extension>
>
>
> So if you attended transfer to 8000, you will park them at the next
> available slot
> This is missing the blf like fifo has but it will be unique announced
> slots.
>
>
> also
>
> you can add this before any call
>
>  <action application="bind_meta_app" data="1 a s valet_park::test auto in
> 8001 8999"/>
>
> so when on the phone you can dial *1 to park the call to the next free slot
> 8001-8999
>
>
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From fraserredmond at gmail.com  Wed Mar 17 13:10:33 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 20:10:33 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
Message-ID: <c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>

Thanks, I suspect this is what I really needed:

> if you are tying to do billing inside some js script you are using, stop
now, and go learn about cdrs.

I guess there's more for me to un-learn from Asterisk than I'd thought :-(

Cheers,
Fraser




On Wed, Mar 17, 2010 at 7:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i'll try again
>
> progress_media and progress are both the same for your purposes
> you will only have exactly one of the 2 set.  its the exact time it started
> ringing.
>
> for our sake, if we have either of them we will take the one that is not
> zero and call it the progress time.
>
> so you have the created time, the progress time , the answered time and the
> hangup time
>
>
> you always have created time and hangup time but not always progress and
> answered
>
> so talk time is (hangup_time - (answered_time unless its 0)) or if
> answered_time is 0 there is no talk time
> ring time is:
>          (answered time unless its zero otherwise hangup_time) -
> created_time
>
>
> edit conf/autoload_configs/cdr_csv.conf.xml
>
> uncomment:
>
> <param name="debug" value="true"/>
>
> and restart FS
>
>
> now when you hangup all your calls you will see the values you have to work
> with in the cdr
>
> if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
>> way?
>>
>> I've realised since my last email that even having the progress_time
>> doesn't help without the progress_media_time or some way to find when the
>> ringing stopped - getting the server timestamp wouldn't happen until the
>> call ended, at which point I can't split the ring time from the talk time.
>>
>> Surely there can't be something that Asterisk handles well that FreeSwitch
>> doesn't? ;-)
>>
>> I'm looking into the CDR now - might have to parse the log files to get
>> the end time from there :-(
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> progress_media_time is only set if you get a 183
>>> if you get a 180 with no media it uses progress_time instead
>>>
>>> as long as you have one or the other you have the timestamp of when it
>>> started ringing.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Yeah, that's what I had expected, and was why I was confused.
>>>>
>>>> I've done some more tests, and tracked the results a little closer in a
>>>> spreadsheet, and it seems that when I call the extension directly from
>>>> another sipphone the  Caller-Channel-Answered-Time minus
>>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>>> fine.
>>>>
>>>> When I call into an IVR with the javascript dialplan and then create the
>>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>>> that are reported after the call ends are reported on the A-leg's Created &
>>>> Answered.
>>>>
>>>> The good news is that the Progress-Time is reported on when the B-leg
>>>> started ringing, so I can know when the call started ringing.
>>>> The bad news is that the Progress-Media-Time is always blank
>>>>
>>>> I can take the Progress-Time and compare it to the system-clock, and
>>>> that should generally be accurate to a second or so, which is better than
>>>> nothing, but it'd be nice to do it properly.
>>>>
>>>> Any other ideas? Is there any channel-variables I can set that would be
>>>> worth playing with? I'm currently only setting:
>>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> even better you have the progress and progress_media timestamps too
>>>>> so you can measure from the instance you got the first ringing
>>>>> indication
>>>>> so then you can also measure how long it took to start ringing too
>>>>> (PDD)
>>>>>
>>>>>
>>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>>
>>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>>> everything, except for tracking one important metric: How long the phone
>>>>>> rang before an agent picked it up.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 13:25:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 13:25:06 -0700
Subject: [Freeswitch-users] Newbie question on freeswitch
In-Reply-To: <8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>
References: <130309.20630.qm@web76211.mail.sg1.yahoo.com>
	<8a19bf2e1003170851o241b8330u14f89a8ddc58ba5@mail.gmail.com>
Message-ID: <87f2f3b91003171325q22cc746dw58675d0cb49b10d2@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:51 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Adjust the dial plan to call extension  8000. The default Dial plan like
> below
>
> <condition field="destination_number" expression="^(10[01][0-9])$">
>
> change to
>
> <condition field="destination_number" expression="^(*[1-8]*0[01][0-9])$">
>

Careful! This causes LOTS of different dialed numbers to match. You might
try this instead:
<condition field="destination_number" expression="^(8000|10[01][0-9])$">

This subject is covered nicely in the newbie article:
http://bit.ly/EpVrv

-MC
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From infos at madovsky.org  Wed Mar 17 13:26:05 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 16:26:05 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
	<191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>
Message-ID: <14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>

Sorry Anthony I never made an official update in C language with diff patch
so the change is at the line 620, I commented out the original line

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 3:55 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  could you get a svn diff of the switch_utils.c so we can  see what the change was?




  On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

    Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
    it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

    I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

    Regards

    Franck
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 16, 2010 4:35 PM
      Subject: Re: [Freeswitch-users] voicemail options


      ok, I tried 50 times to leave a message,
      it worked only one time.
      It seems that sendmail needs to be free to send the email....

        ----- Original Message ----- 
        From: devel at thom.fr.eu.org 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Tuesday, March 16, 2010 3:59 PM
        Subject: Re: [Freeswitch-users] voicemail options


        Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

        I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



        Fran?ois





        De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
        Envoy? : mardi 16 mars 2010 20:30
        ? : freeswitch-users at lists.freeswitch.org
        Objet : Re: [Freeswitch-users] voicemail options



        are you running FS as root or a regular user?
        maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



        On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

        I noticed that sendmail makes a segfault when FS is trying to send email

        and sendmail has some emails locked in folder queue....



        Regards



        Franck


        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org




        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



------------------------------------------------------------------------


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  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Wed Mar 17 13:34:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 13:34:16 -0700
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
Message-ID: <87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>

On Wed, Mar 17, 2010 at 1:10 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks, I suspect this is what I really needed:
>
>
> > if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
> I guess there's more for me to un-learn from Asterisk than I'd thought :-(
>

Yes, there probably is. FreeSWITCH is definitely a paradigm shift, but once
you realize the power under the hood you'll never go back to Asterisk.

If you *really* want to get fancy you can use mod_xml_cdr and get a true 3D
picture of your call. You can even use it to post the XML CDRs to an
external web server and do all your billing offline. :D

You might want to join #freeswitch on irc.freenode.net and rub elbows with
the other Asterisk refugees who've made the transition.

-MC
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From anthony.minessale at gmail.com  Wed Mar 17 13:37:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 15:37:57 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705>
	<191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com>
	<14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
Message-ID: <191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>

FYI if you made the change in your working copy you could generate a patch
with

svn diff > patch.txt

After looking at your change, the problem is that many people depend on this
working the way it currently is.
Would you consider making a script that mimics the current behavior of a
mailer app and read the file from stdin and deliver the subsequent file so
it is backwards-compatible with the way everything already works?


On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

>  Sorry Anthony I never made an official update in C language with diff
> patch
> so the change is at the line 620, I commented out the original line
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 17, 2010 3:55 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> could you get a svn diff of the switch_utils.c so we can  see what the
> change was?
>
>
>
> On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Finally I succeed to send voicemail with sendmail through a little php
>> script (could be Perl also)
>> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
>> doesn't exist.
>>
>> I attached my php script and switch_utils.c. Not sure if the FS developers
>> will accept my modification
>>
>> Regards
>>
>> Franck
>>
>> ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 16, 2010 4:35 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>> ok, I tried 50 times to leave a message,
>> it worked only one time.
>> It seems that sendmail needs to be free to send the email....
>>
>>
>> ----- Original Message -----
>> *From:* devel at thom.fr.eu.org
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Tuesday, March 16, 2010 3:59 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options
>>
>>  Are you running debian, I have the same problem on debian x64 (already
>> mentioned here
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
>> and cannot fix it.
>>
>> I don?t know if it?s a priviledge problem, as I got the same results
>> running FS with or without ?u.
>>
>>
>>
>> Fran?ois
>>
>>
>>
>>
>>
>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
>> Minessale
>> *Envoy? :* mardi 16 mars 2010 20:30
>> *? :* freeswitch-users at lists.freeswitch.org
>> *Objet :* Re: [Freeswitch-users] voicemail options
>>
>>
>>
>> are you running FS as root or a regular user?
>> maybe there is a permission or resource limitation problem preventing it
>> from getting the larger stack space necessary?
>>
>>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> I noticed that sendmail makes a segfault when FS is trying to send email
>>
>> and sendmail has some emails locked in folder queue....
>>
>>
>>
>> Regards
>>
>>
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Wed Mar 17 13:55:55 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 16:55:55 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <1BB02984444E477196CA8761047E79C9@MOBILEE1705>

ok thanks for your help. I will do it next time.
since 3 days I tried hundred times with bash, mailx, mail, same result, somtimes it works sometimes segmnentation fault
with php way it works 100% perfectly now...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


            _______________________________________________
            FreeSWITCH-users mailing list
            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org




            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



--------------------------------------------------------------------


            _______________________________________________
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            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org


        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From fraserredmond at gmail.com  Wed Mar 17 15:23:12 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 22:23:12 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com> 
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com> 
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
Message-ID: <c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>  If you *really* want to get fancy you can use mod_xml_cdr and get a true
> 3D picture of your call. You can even use it to post the XML CDRs to an
> external web server and do all your billing offline. :D
>


Yes, I was planning on using mod_xml_cdr to pass the info to our webserver
to update the database, once I know that mod_cdr_csv is getting the right
data.... which it still isn't  :-(

Much the same problem as before - in mod_cdr_csv's logs the start-time and
answered-time aren't what I expect. I'm gradually getting a better handle on
whats happening...

The 4 significant events in my dialplan are:

ivr-started  (I dial a switchboard number, and enter the extn I want to be
connected to)
call-started  (I'm connected to that extn)
call-ended  (the call with the extn ends)
ivr-ended  (I may return to the extn, or end at the same time as the call)

Master.csv contains 2 rows, both with from and to numbers matching the
caller and callee appropriately.

It seems like the first/blind_transfer row is not picking up the info from
the call that it ought to, and some of it is being recorded against the
second/main ivr row and then lost when it gets new data.

First row has:
start = call-started
answer = empty
end = call-started (probably actually call-connected)
hangup_cause = BLIND_TRANSFER
bleg uuid = empty

Second row (also appears in the separate csv file for the caller's
extension) has
start = ivr-started
answer = ivr-started
end = ivr-ended
hangup_cause = NORMAL_CLEARING
bleg uuid = present

Cheers,
Fraser
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From anthony.minessale at gmail.com  Wed Mar 17 15:37:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Mar 2010 17:37:56 -0500
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com>
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com>
	<c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com>
Message-ID: <191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>

you might want to turn on B-leg logging

On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>  If you *really* want to get fancy you can use mod_xml_cdr and get a true
>> 3D picture of your call. You can even use it to post the XML CDRs to an
>> external web server and do all your billing offline. :D
>>
>
>
> Yes, I was planning on using mod_xml_cdr to pass the info to our webserver
> to update the database, once I know that mod_cdr_csv is getting the right
> data.... which it still isn't  :-(
>
> Much the same problem as before - in mod_cdr_csv's logs the start-time and
> answered-time aren't what I expect. I'm gradually getting a better handle on
> whats happening...
>
> The 4 significant events in my dialplan are:
>
> ivr-started  (I dial a switchboard number, and enter the extn I want to be
> connected to)
> call-started  (I'm connected to that extn)
> call-ended  (the call with the extn ends)
> ivr-ended  (I may return to the extn, or end at the same time as the call)
>
> Master.csv contains 2 rows, both with from and to numbers matching the
> caller and callee appropriately.
>
> It seems like the first/blind_transfer row is not picking up the info from
> the call that it ought to, and some of it is being recorded against the
> second/main ivr row and then lost when it gets new data.
>
> First row has:
> start = call-started
> answer = empty
> end = call-started (probably actually call-connected)
> hangup_cause = BLIND_TRANSFER
> bleg uuid = empty
>
> Second row (also appears in the separate csv file for the caller's
> extension) has
> start = ivr-started
> answer = ivr-started
> end = ivr-ended
> hangup_cause = NORMAL_CLEARING
> bleg uuid = present
>
> Cheers,
> Fraser
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Wed Mar 17 16:11:31 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 17 Mar 2010 23:11:31 +0000
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com> 
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com> 
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com> 
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com> 
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com> 
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com> 
	<c32ee1cf1003171310h6b9aba77we258ccc5f9fdb026@mail.gmail.com> 
	<87f2f3b91003171334w61ba3c49hb00a8a00d460264f@mail.gmail.com> 
	<c32ee1cf1003171523s46a043k3fb6346d5a3330e3@mail.gmail.com> 
	<191c3a031003171537v53d9171fyd20df3d10d18e32@mail.gmail.com>
Message-ID: <c32ee1cf1003171611i3c40fb2cte27b84ec920d7016@mail.gmail.com>

lol - yeah, I guess that would help!  /smacks head against nearby wall/

I'm in the UK and am heading to bed now - I'll sleep much better thanks to
those 8 words from you :-)

Thanks a bunch!

Cheers,
Fraser


On Wed, Mar 17, 2010 at 10:37 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you might want to turn on B-leg logging
>
> On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>>
>> It seems like the first/blind_transfer row is not picking up the info from
>> the call that it ought to, and some of it is being recorded against the
>> second/main ivr row and then lost when it gets new data.
>>
>
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From jmesquita at freeswitch.org  Wed Mar 17 16:37:01 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 17 Mar 2010 20:37:01 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003171251.57460.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003170049.06565.robert.dyck@shaw.ca>
	<ebbc2d551003170501s5da030d7y84d74cb237c19af0@mail.gmail.com>
	<201003171251.57460.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>

No, not at all. Do a lsof -i or a netstat. You should be able to see it all
there. Oh! Wait! Have you made a "make install" on freeswitch trunk?

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> I can see a process for fscomm. Should I be seeing a process for the core.
>
> When starting fscomm it reports -
> Initializing globals...
>
> Initializing core...
>
> Everything OK, Entering runtime loop ...
>
> BTW, not Windows but Fedora 11, KDE flavour
>
> Thanks for looking into this
> RD
>
>
> On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > Rob, I believe that the core has failed to load. I have not added any
> type
> > of check on the code level for those cases yet as I believed it would
> never
> > really fail to load. Those cases have been more frequent than what I
> > originally expected so I will add that to code today (hopefully later
> > tonight).
> >
> > Meanwhile, what you can do is check if the ports are open or not on the
> OS.
> > I am not sure how to do that on Windows. Or, maybe we already have logs
> on
> > the log folder?
> >
> > Regards,
> > Jo?o Mesquita
> >
> > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > Is there something wrong with my build?
> > >
> > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > > [Socket
> > > Connection Error]" with fscomm running.
> > >
> > > RD
> > >
> > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > With FSComm running you can use fs_cli to try pa list. If u can't
> list
> > >
> > > the
> > >
> > > > devices that would be the problem. I'm going to try to dig into this
> > > > further on 1 of my desktops w/ pulseaudio.
> > > >
> > > > -- W
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Wed Mar 17 18:26:33 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 21:26:33 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <4F6DDBC6F1C04B45A6B70316C8D9FC66@MOBILEE1705>


>Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent >file so it is backwards-compatible with the way everything already works?

yes I can do it...
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
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            iax:guest at conference.freeswitch.org/888
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--------------------------------------------------------------------


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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
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  iax:guest at conference.freeswitch.org/888
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  pstn:+19193869900



------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From robert.dyck at shaw.ca  Wed Mar 17 18:46:17 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Wed, 17 Mar 2010 18:46:17 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171251.57460.robert.dyck@shaw.ca>
	<ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
Message-ID: <201003171846.19857.robert.dyck@shaw.ca>

I couldn't remember if I had done an install so I ran it. It didn't make any 
difference though.

Some observations
Freeswitch is not installed in my usual path
Fscomm is not installed by make install
Fs_cli only works if I start freeswitch first
Pa list ( as you suggested earlier ) is not a valid command

Here is the output of lsof -i
fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
>192.168.1.99:nat-pmp 
fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk 
fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk (LISTEN)
fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP localhost.localdomain:oa-
system (LISTEN)

I could not find anything in the netstat output that pertained to freeswitch or 
fscomm.


On Wednesday 17 March 2010 04:37:01 pm Jo?o Mesquita wrote:
> No, not at all. Do a lsof -i or a netstat. You should be able to see it all
> there. Oh! Wait! Have you made a "make install" on freeswitch trunk?
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > I can see a process for fscomm. Should I be seeing a process for the
> > core.
> > 
> > When starting fscomm it reports -
> > Initializing globals...
> > 
> > Initializing core...
> > 
> > Everything OK, Entering runtime loop ...
> > 
> > BTW, not Windows but Fedora 11, KDE flavour
> > 
> > Thanks for looking into this
> > RD
> > 
> > On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > > Rob, I believe that the core has failed to load. I have not added any
> > 
> > type
> > 
> > > of check on the code level for those cases yet as I believed it would
> > 
> > never
> > 
> > > really fail to load. Those cases have been more frequent than what I
> > > originally expected so I will add that to code today (hopefully later
> > > tonight).
> > > 
> > > Meanwhile, what you can do is check if the ports are open or not on the
> > 
> > OS.
> > 
> > > I am not sure how to do that on Windows. Or, maybe we already have logs
> > 
> > on
> > 
> > > the log folder?
> > > 
> > > Regards,
> > > Jo?o Mesquita
> > > 
> > > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > > Is there something wrong with my build?
> > > > 
> > > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error Connecting
> > > > [Socket
> > > > Connection Error]" with fscomm running.
> > > > 
> > > > RD
> > > > 
> > > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > > With FSComm running you can use fs_cli to try pa list. If u can't
> > 
> > list
> > 
> > > > the
> > > > 
> > > > > devices that would be the problem. I'm going to try to dig into
> > > > > this further on 1 of my desktops w/ pulseaudio.
> > > > > 
> > > > > -- W
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > > > http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From infos at madovsky.org  Wed Mar 17 19:49:37 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 17 Mar 2010 22:49:37 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <2D98E43C1BA344E892A250B33927991E@MOBILEE1705>

After dozen of test to correct my php script to be compatible without my patch I noticed
that the real problem comes from "cat" command and not sendmail. I don't know
why but sometimes (very often) the cat  can't write and make a broken pipe, that
of course make sendmail a segmentation fault. So for now I'm locked to use
the patched switch_utils.
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


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            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
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            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



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      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



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      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
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  pstn:+19193869900



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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From cucku.cucku at yahoo.com.vn  Wed Mar 17 19:51:45 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 18 Mar 2010 10:51:45 +0800 (SGT)
Subject: [Freeswitch-users] [roblem with reloadxml command
Message-ID: <585528.94129.qm@web76210.mail.sg1.yahoo.com>

Hi List

i am new to freeswitch, i do the reloadxml in Freeswitch console and get the error:

freeswitch at internal> reloadxml
2010-03-17 15:55:22.694226 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/internal/*.xml (No such file or directory)
+OK [Success]


Please help
Ha`



      __________________________________________________________
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From brian at freeswitch.org  Wed Mar 17 19:56:56 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Mar 2010 21:56:56 -0500
Subject: [Freeswitch-users] [roblem with reloadxml command
In-Reply-To: <585528.94129.qm@web76210.mail.sg1.yahoo.com>
References: <585528.94129.qm@web76210.mail.sg1.yahoo.com>
Message-ID: <0E28DEDC-90BD-414A-A56A-46DFF0060743@freeswitch.org>

You can ignore this.  It means you have nothing to include into the config in that directory.

/b

On Mar 17, 2010, at 9:51 PM, false wrote:

> Hi List
> 
> i am new to freeswitch, i do the reloadxml in Freeswitch console and get the error:
> 
> freeswitch at internal> reloadxml
> 2010-03-17 15:55:22.694226 [ERR] switch_xml.c:1297 Couldnt open /usr/local/freeswitch/conf/dialplan/internal/*.xml (No such file or directory)
> +OK [Success]
> 
> 
> Please help
> Ha`
> 

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From gkuri at ieee.org  Wed Mar 17 20:47:20 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 17 Mar 2010 20:47:20 -0700
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support [Resolved]
Message-ID: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>

We resolved the issue with Broadsoft SCA and inbound phone calls as it
relates to this thread.

If you're using a domain name rather than IP address for $${domain} in
vars.conf, you need to set the sip_invite_domain on the bridge app in
the dialplan, otherwise the INVITEs from FreeSWITCH end up being from
the IP address, which screws up SCA.

For example, we set sip_invite_domain=$${domain} on the bridge app for
local extensions in the dialplan ...

     <action application="bridge"
data="{sip_invite_domain=$${domain}}user/${dialed_extension}@${domain_name}"/>

Shouldn't this be part of the default config in SVN ?

Cheers,
Gabe


On Mon, Mar 1, 2010 at 11:01 PM, Brian West <brian at freeswitch.org> wrote:
> You have to watch phones that might do the switch on you... like resolve the IP then do the subscribe incorrectly... that could happen.
>
> Verify the subscribe packet...
>
> /b
>
> On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:
>
>> I've torn my hair out for hours and I still can't get it to work with
>> the domain name set everywhere <sigh> ...
>>
>> I can get it to work if I hardcode the IP address on the phones and in
>> all the config files, but I can't get the domain name to work on
>> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
>> I'd think it wouldn't work on outgoing calls either, but that seems to
>> work fine.
>>
>> Any other ideas? Might it have anything to do with the incoming call
>> coming in from the PSTN and going through the external profile?
>>
>> Cheers,
>> Gabe
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From jmesquita at freeswitch.org  Wed Mar 17 21:17:51 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 18 Mar 2010 01:17:51 -0300
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003171846.19857.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171251.57460.robert.dyck@shaw.ca>
	<ebbc2d551003171637x1d5e1f3duf203bea1fa118d36@mail.gmail.com>
	<201003171846.19857.robert.dyck@shaw.ca>
Message-ID: <ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>

I think that the problem is that the core is not able to find the
modules....

Try installing the mods on /usr/local/freeswitch/mods. My guess is that
everything will load just fine.

Regards,
Jo?o Mesquita


On Wed, Mar 17, 2010 at 10:46 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:

> I couldn't remember if I had done an install so I ran it. It didn't make
> any
> difference though.
>
> Some observations
> Freeswitch is not installed in my usual path
> Fscomm is not installed by make install
> Fs_cli only works if I start freeswitch first
> Pa list ( as you suggested earlier ) is not a valid command
>
> Here is the output of lsof -i
> fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
> >192.168.1.99:nat-pmp
> fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk
> fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk
> (LISTEN)
> fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP
> localhost.localdomain:oa-
> system (LISTEN)
>
> I could not find anything in the netstat output that pertained to
> freeswitch or
> fscomm.
>
>
> On Wednesday 17 March 2010 04:37:01 pm Jo?o Mesquita wrote:
> > No, not at all. Do a lsof -i or a netstat. You should be able to see it
> all
> > there. Oh! Wait! Have you made a "make install" on freeswitch trunk?
> >
> > Regards,
> > Jo?o Mesquita
> >
> > On Wed, Mar 17, 2010 at 4:51 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > > I can see a process for fscomm. Should I be seeing a process for the
> > > core.
> > >
> > > When starting fscomm it reports -
> > > Initializing globals...
> > >
> > > Initializing core...
> > >
> > > Everything OK, Entering runtime loop ...
> > >
> > > BTW, not Windows but Fedora 11, KDE flavour
> > >
> > > Thanks for looking into this
> > > RD
> > >
> > > On Wednesday 17 March 2010 05:01:15 am Jo?o Mesquita wrote:
> > > > Rob, I believe that the core has failed to load. I have not added any
> > >
> > > type
> > >
> > > > of check on the code level for those cases yet as I believed it would
> > >
> > > never
> > >
> > > > really fail to load. Those cases have been more frequent than what I
> > > > originally expected so I will add that to code today (hopefully later
> > > > tonight).
> > > >
> > > > Meanwhile, what you can do is check if the ports are open or not on
> the
> > >
> > > OS.
> > >
> > > > I am not sure how to do that on Windows. Or, maybe we already have
> logs
> > >
> > > on
> > >
> > > > the log folder?
> > > >
> > > > Regards,
> > > > Jo?o Mesquita
> > > >
> > > > On Wed, Mar 17, 2010 at 4:49 AM, Rob Dyck <robert.dyck at shaw.ca>
> wrote:
> > > > > Is there something wrong with my build?
> > > > >
> > > > > fs_cli yields "[ERROR] libs/esl/fs_cli.c:1181 main() Error
> Connecting
> > > > > [Socket
> > > > > Connection Error]" with fscomm running.
> > > > >
> > > > > RD
> > > > >
> > > > > On Tuesday 16 March 2010 02:36:28 pm William Suffill wrote:
> > > > > > With FSComm running you can use fs_cli to try pa list. If u can't
> > >
> > > list
> > >
> > > > > the
> > > > >
> > > > > > devices that would be the problem. I'm going to try to dig into
> > > > > > this further on 1 of my desktops w/ pulseaudio.
> > > > > >
> > > > > > -- W
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yehavi.bourvine at gmail.com  Wed Mar 17 22:03:26 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 18 Mar 2010 07:03:26 +0200
Subject: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA
	Support [Resolved]
In-Reply-To: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>
References: <8b1c9cda1003172047g33ab3ebcocc6a143c272bb050@mail.gmail.com>
Message-ID: <d361ca7b1003172203x587215fcr9c7bb4a05f51698b@mail.gmail.com>

Thanks! You've done my day...

I am using IPs in order to make SCA working, but this prevents Polycom to do
TLS (it need names). Now both can be made working.

                     Thanks again, __Yehavi:

2010/3/18 Gabriel Kuri <gkuri at ieee.org>

> We resolved the issue with Broadsoft SCA and inbound phone calls as it
> relates to this thread.
>
> If you're using a domain name rather than IP address for $${domain} in
> vars.conf, you need to set the sip_invite_domain on the bridge app in
> the dialplan, otherwise the INVITEs from FreeSWITCH end up being from
> the IP address, which screws up SCA.
>
> For example, we set sip_invite_domain=$${domain} on the bridge app for
> local extensions in the dialplan ...
>
>     <action application="bridge"
> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
> ${domain_name}"/>
>
> Shouldn't this be part of the default config in SVN ?
>
> Cheers,
> Gabe
>
>
> On Mon, Mar 1, 2010 at 11:01 PM, Brian West <brian at freeswitch.org> wrote:
> > You have to watch phones that might do the switch on you... like resolve
> the IP then do the subscribe incorrectly... that could happen.
> >
> > Verify the subscribe packet...
> >
> > /b
> >
> > On Mar 1, 2010, at 11:49 PM, Gabriel Kuri wrote:
> >
> >> I've torn my hair out for hours and I still can't get it to work with
> >> the domain name set everywhere <sigh> ...
> >>
> >> I can get it to work if I hardcode the IP address on the phones and in
> >> all the config files, but I can't get the domain name to work on
> >> incoming calls, I'm at a total loss. If I had a mismatch somewhere,
> >> I'd think it wouldn't work on outgoing calls either, but that seems to
> >> work fine.
> >>
> >> Any other ideas? Might it have anything to do with the incoming call
> >> coming in from the PSTN and going through the external profile?
> >>
> >> Cheers,
> >> Gabe
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Mar 17 22:37:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:37:33 -0700
Subject: [Freeswitch-users] How long did the phone ring?
In-Reply-To: <191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
References: <c32ee1cf1003161521x26f0699aybd46a281edddaaa0@mail.gmail.com>
	<85185EF0-9ACC-4589-91DB-28633FB9D845@freeswitch.org>
	<191c3a031003170957u6e8c110ep419bf5d90529911a@mail.gmail.com>
	<c32ee1cf1003171017o1f4dff6cjd90ab0c5aed59e64@mail.gmail.com>
	<191c3a031003171152u4fa65461hd7a8a30ecae3c5f7@mail.gmail.com>
	<c32ee1cf1003171218y362b158end0ceeda726464703@mail.gmail.com>
	<191c3a031003171241t35c492bboa44c907852bfb82d@mail.gmail.com>
Message-ID: <87f2f3b91003172237l57351ae1g53ca346ce150f6f5@mail.gmail.com>

FYI this is now on the wiki:
http://wiki.freeswitch.org/wiki/CDR#Calculating_Various_Time_Values_For_A_Call

-MC

On Wed, Mar 17, 2010 at 12:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i'll try again
>
> progress_media and progress are both the same for your purposes
> you will only have exactly one of the 2 set.  its the exact time it started
> ringing.
>
> for our sake, if we have either of them we will take the one that is not
> zero and call it the progress time.
>
> so you have the created time, the progress time , the answered time and the
> hangup time
>
>
> you always have created time and hangup time but not always progress and
> answered
>
> so talk time is (hangup_time - (answered_time unless its 0)) or if
> answered_time is 0 there is no talk time
> ring time is:
>          (answered time unless its zero otherwise hangup_time) -
> created_time
>
>
> edit conf/autoload_configs/cdr_csv.conf.xml
>
> uncomment:
>
> <param name="debug" value="true"/>
>
> and restart FS
>
>
> now when you hangup all your calls you will see the values you have to work
> with in the cdr
>
> if you are tying to do billing inside some js script you are using, stop
> now, and go learn about cdrs.
>
>
>
>
>
>
> On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
>> way?
>>
>> I've realised since my last email that even having the progress_time
>> doesn't help without the progress_media_time or some way to find when the
>> ringing stopped - getting the server timestamp wouldn't happen until the
>> call ended, at which point I can't split the ring time from the talk time.
>>
>> Surely there can't be something that Asterisk handles well that FreeSwitch
>> doesn't? ;-)
>>
>> I'm looking into the CDR now - might have to parse the log files to get
>> the end time from there :-(
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> progress_media_time is only set if you get a 183
>>> if you get a 180 with no media it uses progress_time instead
>>>
>>> as long as you have one or the other you have the timestamp of when it
>>> started ringing.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <
>>> fraserredmond at gmail.com> wrote:
>>>
>>>> Yeah, that's what I had expected, and was why I was confused.
>>>>
>>>> I've done some more tests, and tracked the results a little closer in a
>>>> spreadsheet, and it seems that when I call the extension directly from
>>>> another sipphone the  Caller-Channel-Answered-Time minus
>>>> Caller-Channel-Created-Time  matches the ring time, so that scenario is
>>>> fine.
>>>>
>>>> When I call into an IVR with the javascript dialplan and then create the
>>>> new session and bridge them that way, the Answered-Time and Created-Time,
>>>> that are reported after the call ends are reported on the A-leg's Created &
>>>> Answered.
>>>>
>>>> The good news is that the Progress-Time is reported on when the B-leg
>>>> started ringing, so I can know when the call started ringing.
>>>> The bad news is that the Progress-Media-Time is always blank
>>>>
>>>> I can take the Progress-Time and compare it to the system-clock, and
>>>> that should generally be accurate to a second or so, which is better than
>>>> nothing, but it'd be nice to do it properly.
>>>>
>>>> Any other ideas? Is there any channel-variables I can set that would be
>>>> worth playing with? I'm currently only setting:
>>>> ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
>>>>
>>>> Cheers,
>>>> Fraser
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>>> even better you have the progress and progress_media timestamps too
>>>>> so you can measure from the instance you got the first ringing
>>>>> indication
>>>>> so then you can also measure how long it took to start ringing too
>>>>> (PDD)
>>>>>
>>>>>
>>>>> On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> Call start answer time minus call start time = ring time in the CDR
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
>>>>>>
>>>>>> > I'm converting a call-center app from Asterisk to FreeSwitch (using
>>>>>> xml and javascript dialplans) and I think I've worked out how to do nearly
>>>>>> everything, except for tracking one important metric: How long the phone
>>>>>> rang before an agent picked it up.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:+19193869900
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 22:46:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:46:47 -0700
Subject: [Freeswitch-users] freeswitch to cisco gateway
In-Reply-To: <2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
References: <3E1709744536417C877AB5952395DAED@MOBILEE1705>
	<ec2ce17e1003162130s542e3a13wa140d750eea6c62@mail.gmail.com>
	<EEAD83A5-EDDF-4200-AB8B-3EC9289C61AB@freeswitch.org>
	<2303B6A98F994F18BB4B86FD957A4022@MOBILEE1705>
Message-ID: <87f2f3b91003172246u4d8b81a1r29d7aef0dd0da51@mail.gmail.com>

On Wed, Mar 17, 2010 at 8:05 AM, Madovsky <infos at madovsky.org> wrote:

> So why not GPL or Gnu Affero public licence ?
>
> Long story short: the FS devs wanted a different kind of freedom than that
which is offered by the GPL. The GPL is "viral" in that it forces everything
it touches to by GPL-compatible. This kind of force-fed freedom doesn't
always fit well with software that is technically a library (libfreeswitch)
that can be compiled/embedded in other projects.

Bottom line: no OSS license can be all things to all projects. Tony chose
the best one available for his specific project.
-MC
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From msc at freeswitch.org  Wed Mar 17 22:48:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 22:48:41 -0700
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
In-Reply-To: <6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705>
	<6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
Message-ID: <87f2f3b91003172248t33e5ff00nd95dc1e2d8c826c3@mail.gmail.com>

Evidently Google doesn't have 40 million Gizmo users complaining about how
they broke their INVITEs...
-MC

On Wed, Mar 17, 2010 at 10:15 AM, Brian West <brian at freeswitch.org> wrote:

> Sure... I would love that.
>
> /b
>
> On Mar 17, 2010, at 12:11 PM, Madovsky wrote:
>
> But Brian,
>
> are you ready to answer to 40 millions users if  the bug is not resolved ?
> :D
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Mar 17 23:02:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 23:02:17 -0700
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
	<8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
Message-ID: <87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>

On Wed, Mar 17, 2010 at 11:54 AM, Aloysius Lloyd
<lloyd.aloysius at gmail.com>wrote:

>  <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
> mber}"/>
>
> I setup the above dial plan. The call to extension works perfect. But It
> never call to the ${followme_neumber}.
>
> How can I fix this issue.
>
> Thanks
> Lloyd
>
> You might need to set this variable to true prior to your bridge:
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_continue_on_timeout

-MC
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From infos at madovsky.org  Wed Mar 17 23:02:18 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 02:02:18 -0400
Subject: [Freeswitch-users] Skype for SIP - No SILK support, yet...
References: <B10B71FC95A641ABB79B8EA2C40BBCA1@MOBILEE1705><6FB4A329-5B5F-4C81-B216-8639EB3E08E9@freeswitch.org>
	<87f2f3b91003172248t33e5ff00nd95dc1e2d8c826c3@mail.gmail.com>
Message-ID: <F819F014EE9B42D29D4837EDA1189C26@MOBILEE1705>

maybe it's to technical to make users understand ;)
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 1:48 AM
  Subject: Re: [Freeswitch-users] Skype for SIP - No SILK support, yet...


  Evidently Google doesn't have 40 million Gizmo users complaining about how they broke their INVITEs...
  -MC


  On Wed, Mar 17, 2010 at 10:15 AM, Brian West <brian at freeswitch.org> wrote:

    Sure... I would love that.


    /b


    On Mar 17, 2010, at 12:11 PM, Madovsky wrote:


      But Brian,

      are you ready to answer to 40 millions users if  the bug is not resolved ? :D





    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From xengelpublicx at gmail.com  Wed Mar 17 23:11:33 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 09:11:33 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
Message-ID: <eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>

I need to make a queue for the Secretary, in which one secretary. But
if the Secretary is not there, I would like to be able to intercept
calls to other users. This can not be done even if the intercept will
be a user to the agent queue?

On Wed, Mar 17, 2010 at 10:11 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Mar 17, 2010 at 5:22 AM, Vladimir Elizarov <xengelpublicx at gmail.com>
> wrote:
>>
>> Yes
>>
>
> You can't. The FIFO system isn't transferring the call to the agent's phone.
> Rather, it is ringing the agent's phone and when the agent picks up, it then
> bridges the caller to the agent.
>
> What exactly are you trying to accomplish? Just curious. By the way you can
> possibly resolve this issue by create an "agent access" extension. The idea
> is simple: User A picks up and dials an extension; that extension then
> connects User A to the longest waiting caller in the FIFO queue.

I need to make a queue for the Secretary. In this stage only one
secretary. But if the Secretary is not there, I would like to be able
to intercept input calls in fifo queue to other users. This can not be
done even if the intercept will be a user to the agent queue?
>
> Use this example to create your "agent access" extension:
> http://wiki.freeswitch.org/wiki/Mod_fifo#Take_a_caller_out_of_a_FIFO_queue
>
> The net result is probably the same: a third phone connects to the caller in
> queue. If that caller is the only one in the queue then the agent's phone
> that is ringing will stop ringing immediately.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov



From msc at freeswitch.org  Wed Mar 17 23:59:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Mar 2010 23:59:42 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
Message-ID: <87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>

On Wed, Mar 17, 2010 at 11:11 PM, Vladimir Elizarov <xengelpublicx at gmail.com
> wrote:

> I need to make a queue for the Secretary, in which one secretary. But
> if the Secretary is not there, I would like to be able to intercept
> calls to other users. This can not be done even if the intercept will
> be a user to the agent queue?
>
> Why not just have the receptionist log out of the queue when she's not at
her desk?
-MC
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From robert.dyck at shaw.ca  Thu Mar 18 00:00:05 2010
From: robert.dyck at shaw.ca (Rob Dyck)
Date: Thu, 18 Mar 2010 00:00:05 -0700
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171846.19857.robert.dyck@shaw.ca>
	<ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
Message-ID: <201003180000.07647.robert.dyck@shaw.ca>

After figuring out how to get portaudio to build and install, I thought I would 
get somewhere. Alas, it was not to be.

Attempting to load mod_portaudio put me back where I started - no devices

2010-03-17 23:51:02.007720 [ERR] mod_portaudio.c:980 Cannot find an input 
device
2010-03-17 23:51:02.009147 [ERR] mod_portaudio.c:990 Cannot find an input 
device                                                                        
2010-03-17 23:51:02.009147 [CRIT] switch_loadable_module.c:882 Error Loading 
module /usr/local/freeswitch/mod/mod_portaudio.so                          
**Module load routine returned an error** 

Probably, I can't test tomorrow. It's my day with the grandchildren.

RD

On Wednesday 17 March 2010 09:17:51 pm Jo?o Mesquita wrote:
> I think that the problem is that the core is not able to find the
> modules....
> 
> Try installing the mods on /usr/local/freeswitch/mods. My guess is that
> everything will load just fine.
> 
> Regards,
> Jo?o Mesquita
> 
> On Wed, Mar 17, 2010 at 10:46 PM, Rob Dyck <robert.dyck at shaw.ca> wrote:
> > I couldn't remember if I had done an install so I ran it. It didn't make
> > any
> > difference though.
> > 
> > Some observations
> > Freeswitch is not installed in my usual path
> > Fscomm is not installed by make install
> > Fs_cli only works if I start freeswitch first
> > Pa list ( as you suggested earlier ) is not a valid command
> > 
> > Here is the output of lsof -i
> > fscomm  642 rdyck   12u  IPv4 230854      0t0  UDP 192.168.1.3:57205-
> > 
> > >192.168.1.99:nat-pmp
> > 
> > fscomm  642 rdyck   24u  IPv4 231766      0t0  UDP 192.168.1.3:italk
> > fscomm  642 rdyck   25u  IPv4 231767      0t0  TCP 192.168.1.3:italk
> > (LISTEN)
> > fscomm  642 rdyck   28u  IPv4 231902      0t0  TCP
> > localhost.localdomain:oa-
> > system (LISTEN)
> > 
> > I could not find anything in the netstat output that pertained to
> > freeswitch or
> > fscomm.



From tculjaga at gmail.com  Thu Mar 18 00:08:04 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 18 Mar 2010 08:08:04 +0100
Subject: [Freeswitch-users] Release Cause in response message
In-Reply-To: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>
References: <65d96fc81003170746x290c2232o3dd0dfe4c8ff8847@mail.gmail.com>
Message-ID: <65d96fc81003180008l2238f9f9vac5545ce605787e5@mail.gmail.com>

any way to pass a disconnect reason vi a respond command ???

anyone?

T.

On Wed, Mar 17, 2010 at 3:46 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello folks,
>
>
> anyone knows how to pass a Release cause/ Disconnect Reason when firing
> respond from a dialplan ... e.g.:
>
> Ive tried with sip_h_Reason=Q.850;cause=27 but it is not working :)
>
>
>
>    <extension name="parseResult">
>       <condition field="destination_number" expression="^parseResult$"/>
>       <condition field="${componentStatus}" expression="^1$"
> break="on-true">
>          <action application="log" data="INFO ######################## CALL
> FAILED REASON 1' ##########\n"/>
>
>          <action application="set" data="sip_h_Reason=Q.850;cause=16"/>
>
>          <action application="respond" data="480 Temporarily Unavailable"/>
>       </condition>
>    </extension>
>
>
> any hint?
>
>
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From xengelpublicx at gmail.com  Thu Mar 18 00:39:59 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 10:39:59 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
Message-ID: <eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>

If she leaves the queue, the queue will not be agents. A user simply
intercepting the call should be able to transfer the call to your
phone when the Secretary is not there.

Agents can intercept a call in the fifo queue?

On Thu, Mar 18, 2010 at 9:59 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Mar 17, 2010 at 11:11 PM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>>
>> I need to make a queue for the Secretary, in which one secretary. But
>> if the Secretary is not there, I would like to be able to intercept
>> calls to other users. This can not be done even if the intercept will
>> be a user to the agent queue?
>>
> Why not just have the receptionist log out of the queue when she's not at
> her desk?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov



From brian at microcomaustralia.com.au  Thu Mar 18 00:56:49 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Thu, 18 Mar 2010 18:56:49 +1100
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>
References: <3c5cf5261003170117h21fdcd41u96e9abd31bebab56@mail.gmail.com>
	<4C8BD46E2AB84D6F85EC0B716FB11B35@cune.pri>
Message-ID: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>

On 17 March 2010 21:57, Russell Mosemann <Russell.Mosemann at cune.org> wrote:
> Make the appropriate changes to the configuration files in /etc/dahdi/. Take a look at the files in /proc/dahdi/.

Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? I tried
copying the file, it doesn't seem to have changed anything...

huey:~# cat /proc/dahdi/1
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

	   1 WCTDM/4/0
	   2 WCTDM/4/1
	   3 WCTDM/4/2
	   4 WCTDM/4/3
huey:~# cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)
fxoks=1
fxoks=2
fxsks=3
fxsks=4

# Global data

loadzone	= au
defaultzone	= au


What process reads this file?
-- 
Brian May <brian at microcomaustralia.com.au>



From maciej.aniserowicz at gmail.com  Thu Mar 18 01:19:30 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Thu, 18 Mar 2010 00:19:30 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003150939l5d33e830gfd317a9a06376544@mail.gmail.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
Message-ID: <1268900370911-4755257.post@n2.nabble.com>


You're right, sorry.

I updated to trunk (rev 17031) and the problem is still here.

Logs produced by newest version: http://pastebin.freeswitch.org/12474. They
contain comments and commands that I used.

Scenario:
1st call is made and is parked. Then 2nd and 3rd calls are:
1) dialed
2) parked
3) bridged with 1st call
4) hanged up

It is OK for now.
Then a 4th call is dialed, parked, bridged and killed. After killing the 4th
call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.


I hope this description and logs help resolve the issue.
Please let me know if you need any more information.

FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From cucku.cucku at yahoo.com.vn  Thu Mar 18 01:31:15 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Thu, 18 Mar 2010 16:31:15 +0800 (SGT)
Subject: [Freeswitch-users] Newbie question on domain name
Message-ID: <554663.90897.qm@web76203.mail.sg1.yahoo.com>

Hi List

i feel strange about default domain in Freswitch:
i read the vars.xml and default domain name is IP address:
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

and i register successfull the 1000, 1002 extension
the i copy 1000.xml to 7000.xml 
then i register 7000 extension but get error

Please help
Ha`

Registrations:
=================================================================================================
Call-ID:??????? 56dc9e359865070c at 192.168.1.6
User:?????????? 1002 at 192.168.1.11
Contact:??????? "user" <sip:1002 at 192.168.1.6:4318;user=phone>
Agent:????????? Grandstream BT110 1.0.7.3
Status:???????? Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
Host:?????????? fs
IP:???????????? 192.168.1.6
Port:?????????? 4318
Auth-User:????? 1002
Auth-Realm:???? 192.168.1.11
MWI-Account:??? 1002 at 192.168.1.11

Call-ID:??????? 8fa759179877592f at 192.168.1.12
User:?????????? 1009 at 192.168.1.11
Contact:??????? "user" <sip:1009 at 192.168.1.12:57386;user=phone>
Agent:????????? Grandstream BT110 1.0.7.3
Status:???????? Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
Host:?????????? fs
IP:???????????? 192.168.1.12
Port:?????????? 57386
Auth-User:????? 1009
Auth-Realm:???? 192.168.1.11
MWI-Account:??? 1009 at 192.168.1.11

=================================================================================================

freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866 Can't find user [7000 at 192.168.1.11]
You must define a domain called '192.168.1.11' in your directory and add a user with the id="7000" attribute
and you must configure your device to use the proper domain in it's authentication credentials.
2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth failure (REGISTER) on sofia profile 'internal' for [7000 at 192.168.1.11] from ip 192.168.1.2




      Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n! T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?. http://vn.messenger.yahoo.com/pingbox/
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From tjardick at vanderkraan.net  Thu Mar 18 01:51:03 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 09:51:03 +0100
Subject: [Freeswitch-users] Newbie question on domain name
In-Reply-To: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
References: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
Message-ID: <B2EE99FE-EDE1-4717-BF06-536F522B1143@vanderkraan.net>

Hi False,

Did you edit the 7000.xml to reflect the changes within the file?

Also did you perform a "reloadxml" to tell freeswitch to reread it's  
configuration?


Regards,

Tj

On 18 Mar 2010, at 09:31, false wrote:

> Hi List
>
> i feel strange about default domain in Freswitch:
> i read the vars.xml and default domain name is IP address:
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
> and i register successfull the 1000, 1002 extension
> the i copy 1000.xml to 7000.xml
> then i register 7000 extension but get error
>
> Please help
> Ha`
>
> Registrations:
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> ======================================================================
> Call-ID:        56dc9e359865070c at 192.168.1.6
> User:           1002 at 192.168.1.11
> Contact:        "user" <sip:1002 at 192.168.1.6:4318;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
> Host:           fs
> IP:             192.168.1.6
> Port:           4318
> Auth-User:      1002
> Auth-Realm:     192.168.1.11
> MWI-Account:    1002 at 192.168.1.11
>
> Call-ID:        8fa759179877592f at 192.168.1.12
> User:           1009 at 192.168.1.11
> Contact:        "user" <sip:1009 at 192.168.1.12:57386;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
> Host:           fs
> IP:             192.168.1.12
> Port:           57386
> Auth-User:      1009
> Auth-Realm:     192.168.1.11
> MWI-Account:    1009 at 192.168.1.11
>
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> = 
> ======================================================================
>
> freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866  
> Can't find user [7000 at 192.168.1.11]
> You must define a domain called '192.168.1.11' in your directory and  
> add a user with the id="7000" attribute
> and you must configure your device to use the proper domain in it's  
> authentication credentials.
> 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth  
> failure (REGISTER) on sofia profile 'internal' for  
> [7000 at 192.168.1.11] from ip 192.168.1.2
>
>
> Truy c?p Mail nhanh h?n!
> Yahoo! khuy?n kh?ch b?n n?ng c?p tr?nh duy?t l?n Internet  
> Explorer 8 m?i, t?i ?u h?a cho Yahoo!. T?i t?i ??y!  
> (Mi?n ph?)_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Thu Mar 18 04:39:46 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 18 Mar 2010 22:39:46 +1100
Subject: [Freeswitch-users] Fscomm unable to detect devices
In-Reply-To: <201003180000.07647.robert.dyck@shaw.ca>
References: <201003141811.58645.robert.dyck@shaw.ca>
	<201003171846.19857.robert.dyck@shaw.ca>
	<ebbc2d551003172117m40e70a17x221244e095154e4f@mail.gmail.com>
	<201003180000.07647.robert.dyck@shaw.ca>
Message-ID: <20100318113946.GA1716@jdc.jasonjgw.net>

Rob Dyck <robert.dyck at shaw.ca> wrote:
> After figuring out how to get portaudio to build and install, I thought I
> would get somewhere. Alas, it was not to be.
> 
> Attempting to load mod_portaudio put me back where I started - no devices
> 
> 2010-03-17 23:51:02.007720 [ERR] mod_portaudio.c:980 Cannot find an input
> device 

I've seen this before; it was a permission problem, solved by adding the
freeswitch user to the audio group so that it could access the ALSA devices.

Whether this is the cause in your case depends on how your operating system
controls access to audio devices.




From rentmycoder at gmail.com  Thu Mar 18 05:21:09 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Thu, 18 Mar 2010 13:21:09 +0100
Subject: [Freeswitch-users] how to dial out through a local sip endpoint -
	mv372 mobile gateway adapter
Message-ID: <50e456911003180521je008a2dme94fcc6901a04b99@mail.gmail.com>

Hi guys,
Freeswitch rocks!
I've registered an Portech mv-372 mobile adapter to FS as an internal
user as userid/extension=5000.
How to dial out through this endpint?

In asterisk I've used Dial(sip/1234567 at 5000) where 1234567 is the
phone no to call...

In FS: bridge(sophia/internal/1234567 at 5000)
mod_dptools: normal_temporery_failure

any ideas how to do the bridge?
or should I setup the adapter as a gateway profile?

Just to make sure the adapter is ok i've tested it using asterisk and
it worked and was set up like this:
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk



From phunk0000 at hotmail.com  Thu Mar 18 05:22:10 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 08:22:10 -0400
Subject: [Freeswitch-users] FS Best practices
Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>

Hey List, still kinda new to this whole deal, looking for some advice.  We
are building a website that needs to interact with FS.  Not looking for
specific instructions just opinions about the best practices for doing
things like click to dial from a web page, calling CDR records to web
applications to display customers stats and general web integration.  Does
this all need to be done using scripting languages like Lua or Perl, or can
you directly integrate FS with a website in some other manner. Again, just
getting into all this, but having a blast learning FS, config and power
rock!  Anyhow, some places or things I should start looking into would be
greatly appreciated.  Thanks List!

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From gmaruzz at celliax.org  Thu Mar 18 05:33:13 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 18 Mar 2010 13:33:13 +0100
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
Message-ID: <7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.? We
> are building a website that needs to interact with FS.? Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.? Does
> this all need to be done using scripting languages like Lua or Perl, or can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.



> getting into all this, but having a blast learning FS, config and power
> rock!? Anyhow, some places or things I should start looking into would be
> greatly appreciated.? Thanks List!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From steve at justfone.com  Thu Mar 18 05:39:10 2010
From: steve at justfone.com (Steven Brown)
Date: Thu, 18 Mar 2010 12:39:10 +0000
Subject: [Freeswitch-users] Re : how to dial out through a local sip
	endpoint - mv372 mobile > gateway adapter (rentmycoder rentmycoder)
Message-ID: <3e6d7b0c1003180539y5d8b300bl4392f72c34129ba3@mail.gmail.com>

Hi,

You can do it with the Portech  registered on FS as a regular endpoint
then  bridge to it
as below or similar

<extension name="gsm">
     <condition field="destination_number" expression="^9(.*)$">
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="bridge" data="sofia/internal/1005^$1@${domain}">

where the Portech registers as 1005 and I prefix the number I wish to
dial with 9

or alternatively just bridge directly to it without having to register

<action application="bridge" data="sofia/default/$1 at 192.168.1.3:5060"/>

Where 192.168.1.3 is the Portech

In both cases you need to configure the Portech Lan to Mobile Table as below

Item  URL                 Call Num
0       192.168.1.2     #

where 192.168.1.2 is my FS box

Steve


> Message: 5
> Date: Thu, 18 Mar 2010 13:21:09 +0100
> From: rentmycoder rentmycoder <rentmycoder at gmail.com>
> Subject: [Freeswitch-users] how to dial out through a local sip
> ? ? ? ?endpoint - ? ? ?mv372 mobile gateway adapter
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
> ? ? ? ?<50e456911003180521je008a2dme94fcc6901a04b99 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi guys,
> Freeswitch rocks!
> I've registered an Portech mv-372 mobile adapter to FS as an internal
> user as userid/extension=5000.
> How to dial out through this endpint?
>
> In asterisk I've used Dial(sip/1234567 at 5000) where 1234567 is the
> phone no to call...
>
> In FS: bridge(sophia/internal/1234567 at 5000)
> mod_dptools: normal_temporery_failure
>
> any ideas how to do the bridge?
> or should I setup the adapter as a gateway profile?
>
> Just to make sure the adapter is ok i've tested it using asterisk and
> it worked and was set up like this:
> http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>
>
>
> ------------------------------
>
> Message: 6
> Date: Thu, 18 Mar 2010 08:22:10 -0400
> From: "Todd" <phunk0000 at hotmail.com>
> Subject: [Freeswitch-users] FS Best practices
> To: "FreeSWITCH-users at lists. freeswitch. org "
> ? ? ? ?<FreeSWITCH-users at lists.freeswitch.org>
> Message-ID: <BLU110-DS11B84F16B12464595A3B07A72B0 at phx.gbl>
> Content-Type: text/plain; charset="us-ascii"
>
> Hey List, still kinda new to this whole deal, looking for some advice. ?We
> are building a website that needs to interact with FS. ?Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration. ?Does
> this all need to be done using scripting languages like Lua or Perl, or can
> you directly integrate FS with a website in some other manner. Again, just
> getting into all this, but having a blast learning FS, config and power
> rock! ?Anyhow, some places or things I should start looking into would be
> greatly appreciated. ?Thanks List!
>
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> End of FreeSWITCH-users Digest, Vol 45, Issue 225
> *************************************************
>



-- 
Steven Brown

email   steve at justfone.com
office   08707706968
mobile 07768755409
fax      07884636663

Justfone - Company Reg. No.  : 3926817

Registered Office : 1-3 Sandgate, Berwick upon Tweed, Northumberland, TD15 1EW

The contents of this e-mail may be privileged and are confidential. It
may not be disclosed to or used by anyone other than the addressee(s),
nor copied in any way. If received in error, please advise sender,
then delete it from your system. Internet email communications are not
secure and therefore Justfone do not accept legal responsibility for
the contents of this message. Any views or opinions presented are
solely those of the author and do not necessarily represent those of
Justfone unless otherwise specifically stated.



From Russell.Mosemann at cune.org  Thu Mar 18 06:23:19 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Thu, 18 Mar 2010 13:23:19 -0000
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
Message-ID: <20100318132319.2A5BA312875@cuneorg-email.cune.pri>

Brian May <brian at microcomaustralia.com.au> said:

> Is /etc/dahdi/system.conf the same as /etc/zaptel.conf?

Just glancing through the file, it looks the same.

> What process reads this file?

/etc/init.d/dahdi

What is in /etc/dahdi/modules?

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From lloyd.aloysius at gmail.com  Thu Mar 18 06:32:05 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 09:32:05 -0400
Subject: [Freeswitch-users] What is the best way implement the follow me
	for a extension
In-Reply-To: <87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>
References: <8a19bf2e1003081433m5613ce75sba62ef433042fbe8@mail.gmail.com>
	<87f2f3b91003081510v6c920f4ahe7c791d2de55e9c8@mail.gmail.com>
	<8a19bf2e1003081524g1b159aa9l8bc9ec1d66ac3e36@mail.gmail.com>
	<8a19bf2e1003171154n28a1ca47jd5139840be828341@mail.gmail.com>
	<87f2f3b91003172302k5ba8d2ddnfcf8be3677616cc9@mail.gmail.com>
Message-ID: <8a19bf2e1003180632h3b453042q7860cc7852421191@mail.gmail.com>

Michael, Thank you. I will try your suggestion.

Thanks
Lloyd

On Thu, Mar 18, 2010 at 2:02 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Mar 17, 2010 at 11:54 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
> > wrote:
>
>>  <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name}|[leg_delay_start=12,leg_timeout=30]sofia/gateway/voipms/${followme_nu
>> mber}"/>
>>
>> I setup the above dial plan. The call to extension works perfect. But It
>> never call to the ${followme_neumber}.
>>
>> How can I fix this issue.
>>
>> Thanks
>> Lloyd
>>
>> You might need to set this variable to true prior to your bridge:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_continue_on_timeout
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 18 06:55:37 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 18 Mar 2010 06:55:37 -0700
Subject: [Freeswitch-users] Newbie question on domain name
In-Reply-To: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
References: <554663.90897.qm@web76203.mail.sg1.yahoo.com>
Message-ID: <4712D923-82FF-45D8-B73C-A8CF3816E2AB@freeswitch.org>

Did you edit 7000.xml? Make sure that you open that file and put  
"7000" in place of "1000".
-MC
Sent from my iPhone

On Mar 18, 2010, at 1:31 AM, false <cucku.cucku at yahoo.com.vn> wrote:

> Hi List
>
> i feel strange about default domain in Freswitch:
> i read the vars.xml and default domain name is IP address:
> <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
>
> and i register successfull the 1000, 1002 extension
> the i copy 1000.xml to 7000.xml
> then i register 7000 extension but get error
>
> Please help
> Ha`
>
> Registrations:
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> ======================================================================
> Call-ID:        56dc9e359865070c at 192.168.1.6
> User:           1002 at 192.168.1.11
> Contact:        "user" <sip:1002 at 192.168.1.6:4318;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:46:13)
> Host:           fs
> IP:             192.168.1.6
> Port:           4318
> Auth-User:      1002
> Auth-Realm:     192.168.1.11
> MWI-Account:    1002 at 192.168.1.11
>
> Call-ID:        8fa759179877592f at 192.168.1.12
> User:           1009 at 192.168.1.11
> Contact:        "user" <sip:1009 at 192.168.1.12:57386;user=phone>
> Agent:          Grandstream BT110 1.0.7.3
> Status:         Registered(UDP)(unknown) EXP(2010-03-18 04:48:08)
> Host:           fs
> IP:             192.168.1.12
> Port:           57386
> Auth-User:      1009
> Auth-Realm:     192.168.1.11
> MWI-Account:    1009 at 192.168.1.11
>
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> === 
> ======================================================================
>
> freeswitch at fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866  
> Can't find user [7000 at 192.168.1.11]
> You must define a domain called '192.168.1.11' in your directory and  
> add a user with the id="7000" attribute
> and you must configure your device to use the proper domain in it's  
> authentication credentials.
> 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth  
> failure (REGISTER) on sofia profile 'internal' for  
> [7000 at 192.168.1.11] from ip 192.168.1.2
>
>
> Truy c?p Mail nhanh h?n!
> Yahoo! khuy?n kh?ch b?n n?ng c?p tr?nh duy?t l?n Internet  
> Explorer 8 m?i, t?i ?u h?a cho Yahoo!. T?i t?i ??y!  
> (Mi?n ph?)
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From moises.silva at gmail.com  Thu Mar 18 07:19:34 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 18 Mar 2010 10:19:34 -0400
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <20100318132319.2A5BA312875@cuneorg-email.cune.pri>
References: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
	<20100318132319.2A5BA312875@cuneorg-email.cune.pri>
Message-ID: <c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>

do svn update ( that will fix some of the errors but NOT the critical one).
The critical one is "failure configuring device ... " and I think that one
is because you did not run "dahdi_cfg" utility.

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

On Thu, Mar 18, 2010 at 9:23 AM, <Russell.Mosemann at cune.org> wrote:

> Brian May <brian at microcomaustralia.com.au> said:
>
> > Is /etc/dahdi/system.conf the same as /etc/zaptel.conf?
>
> Just glancing through the file, it looks the same.
>
> > What process reads this file?
>
> /etc/init.d/dahdi
>
> What is in /etc/dahdi/modules?
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Thu Mar 18 07:38:01 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 10:38:01 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
Message-ID: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>

Hi All,

The following lua script for calculate the call duration. But the issue here
, the duration is calculating from the outbound initiate time.

How to get the call answer time?


session:preAnswer();
-- sleep a second
session:sleep(1000);
--
digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav",
"invalid-digits.wav", "\\d+|\\*");

-- Initiate an outbound call
outSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session)


starttime  = os.time();
freeswitch.consoleLog("info",   "Start Time"..starttime.." \n");

freeswitch.consoleLog("info",   "Before Bridge the Call \n");
freeswitch.bridge(session, outSession);
freeswitch.consoleLog("info",   "After Bridge the Call \n");

endtime = os.time();
freeswitch.consoleLog("info",   "End Time"..endtime.." \n");

billsec = os.difftime(endtime,starttime);
freeswitch.consoleLog("info",   "Blling Sec"..billsec.." \n");

-- hangup
session:hangup();


Thanks
Lloyd
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From msc at freeswitch.org  Thu Mar 18 08:16:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:16:37 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
Message-ID: <87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>

On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> The following lua script for calculate the call duration. But the issue
> here , the duration is calculating from the outbound initiate time.
>
> How to get the call answer time?
>
If you're trying to do billing in-line you might as well stop right now and
read the CDRs page on the wiki. Trying to calculate this from inside the
session is silly.
-MC
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From msc at freeswitch.org  Thu Mar 18 08:29:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:29:24 -0700
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
Message-ID: <87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> > Hey List, still kinda new to this whole deal, looking for some advice.
> We
> > are building a website that needs to interact with FS.  Not looking for
> > specific instructions just opinions about the best practices for doing
> > things like click to dial from a web page, calling CDR records to web
> > applications to display customers stats and general web integration.
> Does
> > this all need to be done using scripting languages like Lua or Perl, or
> can
> > you directly integrate FS with a website in some other manner. Again,
> just
>
> You definitely want to have a look at:
> http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
> pages.
>
>
I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 08:32:54 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 11:32:54 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
Message-ID: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>

Thank you for the suggestion. Yes I went to the direction then Stop. How to
handle Hangup actions for every call?

For Example : I do not want to calculate the duration for all the calls or
execute actions for all the calls. I need calculate for particular calls
only.I tested the following ,* it execute for every calls hangup*. How to
control or How to execute in some conditions.


cdr_csv.conf.xml


<param name="default-template" value="lua"/>

<template name="lua">${lua(onhangup.lua)}</template>


On Thu, Mar 18, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Hi All,
>>
>> The following lua script for calculate the call duration. But the issue
>> here , the duration is calculating from the outbound initiate time.
>>
>> How to get the call answer time?
>>
> If you're trying to do billing in-line you might as well stop right now and
> read the CDRs page on the wiki. Trying to calculate this from inside the
> session is silly.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sos at sokhapkin.dyndns.org  Thu Mar 18 08:46:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 18 Mar 2010 11:46:11 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
Message-ID: <201003181146.11642.sos@sokhapkin.dyndns.org>

<action application="set" data="process_cdr=false"/>

On Thursday 18 March 2010, Aloysius Lloyd wrote:
> Thank you for the suggestion. Yes I went to the direction then Stop. How to
> handle Hangup actions for every call?
> 
> For Example : I do not want to calculate the duration for all the calls or
> execute actions for all the calls. I need calculate for particular calls
> only.I tested the following ,* it execute for every calls hangup*. How to
> control or How to execute in some conditions.
> 
> 
> cdr_csv.conf.xml
> 
> 
> <param name="default-template" value="lua"/>
> 
> <template name="lua">${lua(onhangup.lua)}</template>
> 
> On Thu, Mar 18, 2010 at 11:16 AM, Michael Collins <msc at freeswitch.org>wrote:
> > On Thu, Mar 18, 2010 at 7:38 AM, Aloysius Lloyd 
<lloyd.aloysius at gmail.com>wrote:
> >> Hi All,
> >>
> >> The following lua script for calculate the call duration. But the issue
> >> here , the duration is calculating from the outbound initiate time.
> >>
> >> How to get the call answer time?
> >
> > If you're trying to do billing in-line you might as well stop right now
> > and read the CDRs page on the wiki. Trying to calculate this from inside
> > the session is silly.
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 




From msc at freeswitch.org  Thu Mar 18 08:49:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:49:29 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
Message-ID: <87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>

On Thu, Mar 18, 2010 at 8:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Thank you for the suggestion. Yes I went to the direction then Stop. How to
> handle Hangup actions for every call?
>
> For Example : I do not want to calculate the duration for all the calls or
> execute actions for all the calls. I need calculate for particular calls
> only.I tested the following ,* it execute for every calls hangup*. How to
> control or How to execute in some conditions.
>

Is there a problem with running it on every call? The overhead of a Lua
script is minimal compared to the actual call. I'm just curious what you
gain from selectively running the script.
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 09:05:00 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 12:05:00 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
Message-ID: <8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>

 for example in a 10000 calls only want to run 100 times. Do you think it is
a best practice run the Lua script for 9900 time for no reason.

Lloyd

On Thu, Mar 18, 2010 at 11:49 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 8:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Thank you for the suggestion. Yes I went to the direction then Stop. How
>> to handle Hangup actions for every call?
>>
>> For Example : I do not want to calculate the duration for all the calls or
>> execute actions for all the calls. I need calculate for particular calls
>> only.I tested the following ,* it execute for every calls hangup*. How to
>> control or How to execute in some conditions.
>>
>
> Is there a problem with running it on every call? The overhead of a Lua
> script is minimal compared to the actual call. I'm just curious what you
> gain from selectively running the script.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Mar 18 08:57:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 08:57:37 -0700
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
	<eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
Message-ID: <87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>

On Thu, Mar 18, 2010 at 12:39 AM, Vladimir Elizarov <xengelpublicx at gmail.com
> wrote:

> If she leaves the queue, the queue will not be agents. A user simply
> intercepting the call should be able to transfer the call to your
> phone when the Secretary is not there.
>
I'm sorry but I don't understand. If the receptionist logs out of the queue
then calls that are queued up will go to other agents (if available) or stay
in queue until the receptionist gets back. Also, you can do the "agent
access" extension I mentioned earlier so that non-agents can take callers
out of the queue.
-MC
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From msc at freeswitch.org  Thu Mar 18 09:11:52 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 09:11:52 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
Message-ID: <87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>

On Thu, Mar 18, 2010 at 9:05 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

>  for example in a 10000 calls only want to run 100 times. Do you think it
> is a best practice run the Lua script for 9900 time for no reason.

Do you need to skip processing the CDR altogether for those other 9900
calls?
-MC
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From lloyd.aloysius at gmail.com  Thu Mar 18 09:22:23 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 18 Mar 2010 12:22:23 -0400
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
	<87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
Message-ID: <8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>

No.Need CDR for all calls.

I am thinking in the following way. Is this possible?

Application A - Need CDR values and cdr_csv.conf.xml use *onhangupA.lua*

Application B - Need CDR values and cdr_csv.conf.xml use *onhangupB.lua*

Application C - Need CDR values only.


Thanks
Lloyd

On Thu, Mar 18, 2010 at 12:11 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Mar 18, 2010 at 9:05 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>>  for example in a 10000 calls only want to run 100 times. Do you think it
>> is a best practice run the Lua script for 9900 time for no reason.
>
> Do you need to skip processing the CDR altogether for those other 9900
> calls?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at redbonez.net  Thu Mar 18 09:52:13 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 10:52:13 -0600
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
Message-ID: <021d01cac6bb$5d73e1e0$185ba5a0$@net>

If you are a Python fan you should check out Twisted and
http://code.google.com/p/eventsocket/

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, March 18, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

 

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
wrote:

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.  We
> are building a website that needs to interact with FS.  Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.  Does
> this all need to be done using scripting languages like Lua or Perl, or
can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.

 


I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC

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From phunk0000 at hotmail.com  Thu Mar 18 10:37:23 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 13:37:23 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <021d01cac6bb$5d73e1e0$185ba5a0$@net>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
Message-ID: <BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>

Thanks folks!  I believe we will be doing most of our coding in php.  We are
also going to have independent DB servers.. I was also wondering if putting
the FS database on a DB server is a god idea, or if it should be left on the
FS box.  'Preciate it yall.  Learning is phun

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam
Ford
Sent: Thursday, March 18, 2010 12:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

If you are a Python fan you should check out Twisted and
http://code.google.com/p/eventsocket/

 

-Adam

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, March 18, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

 

On Thu, Mar 18, 2010 at 5:33 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
wrote:

On Thu, Mar 18, 2010 at 1:22 PM, Todd <phunk0000 at hotmail.com> wrote:
> Hey List, still kinda new to this whole deal, looking for some advice.  We
> are building a website that needs to interact with FS.  Not looking for
> specific instructions just opinions about the best practices for doing
> things like click to dial from a web page, calling CDR records to web
> applications to display customers stats and general web integration.  Does
> this all need to be done using scripting languages like Lua or Perl, or
can
> you directly integrate FS with a website in some other manner. Again, just

You definitely want to have a look at:
http://wiki.freeswitch.org/wiki/Event_Socket_Library and related
pages.

 


I agree with Giovanni: The event socket is your friend. You can build your
app server on a separate box from your FS box which makes it easier to
scale, test, replicate...
-MC

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.791 / Virus Database: 271.1.1/2752 - Release Date: 03/18/10
03:33:00

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From brian at freeswitch.org  Thu Mar 18 10:41:59 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 12:41:59 -0500
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
Message-ID: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>

Not sure how much god has to do with this... Unless you're praying your DB doesn't crash.   Kidding aside.  Its really up to you it works both ways.  But I would recommend a dedicated network for interconnection and management and not run that over the public/sip facing interfaces.

/b

On Mar 18, 2010, at 12:37 PM, Todd wrote:

> Thanks folks!  I believe we will be doing most of our coding in php.  We are also going to have independent DB servers?. I was also wondering if putting the FS database on a DB server is a god idea, or if it should be left on the FS box.  ?Preciate it yall.  Learning is phun

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From msc at freeswitch.org  Thu Mar 18 10:43:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 10:43:13 -0700
Subject: [Freeswitch-users] Calculate the Call Duration ( BillSec)
In-Reply-To: <8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>
References: <8a19bf2e1003180738l16771d1ev732bceb4d565f838@mail.gmail.com>
	<87f2f3b91003180816o1fe5fe72wed7752f73139cac3@mail.gmail.com>
	<8a19bf2e1003180832o1c7daa0aue99eafacc2cbb06d@mail.gmail.com>
	<87f2f3b91003180849g6eab0c90se9cac3f878f6eaac@mail.gmail.com>
	<8a19bf2e1003180905n6d9122edj586e60bd8ded6858@mail.gmail.com>
	<87f2f3b91003180911k4a699667g3bbe0ada9f834848@mail.gmail.com>
	<8a19bf2e1003180922l4e09fcdcg9f9916dbc577caf2@mail.gmail.com>
Message-ID: <87f2f3b91003181043h63735e06p80719bdaf816a71a@mail.gmail.com>

On Thu, Mar 18, 2010 at 9:22 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> No.Need CDR for all calls.
>
> I am thinking in the following way. Is this possible?
>
> Application A - Need CDR values and cdr_csv.conf.xml use *onhangupA.lua*
>
> Application B - Need CDR values and cdr_csv.conf.xml use *onhangupB.lua*
>
> Application C - Need CDR values only.
>

What you really need to be using is mod_xml_cdr and posting your CDRs to a
separate server and do all this stuff outside of the FS server. Using
mod_xml_cdr will definitely scale more easily than trying to play around
with crazy logic in the dialplan and CSV CDRs.

http://wiki.freeswitch.org/wiki/Mod_xml_cdr

It's a paradigm shift from CSV CDRs but when you realize just how powerful
it is you will never be satisfied with plain CSV CDRs any more.
-MC
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From jonas.gauffin at gmail.com  Thu Mar 18 10:46:25 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Thu, 18 Mar 2010 18:46:25 +0100
Subject: [Freeswitch-users] Calls
Message-ID: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>

Hello,

Is it possible to list calls that are terminated in IVR's, are parked
or in early state?
"show calls" api just lists established calls, correct?

Best regards,
  Jonas



From phunk0000 at hotmail.com  Thu Mar 18 10:51:28 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 18 Mar 2010 13:51:28 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>	<021d01cac6bb$5d73e1e0$185ba5a0$@net>	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
Message-ID: <BLU110-DS912F0C62DC672E60B3D59A72B0@phx.gbl>

Ha! GOOD, thanks, all the servers are nice and local with gigabit
cards/switches.  Thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 18, 2010 1:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Best practices

 

Not sure how much god has to do with this... Unless you're praying your DB
doesn't crash.   Kidding aside.  Its really up to you it works both ways.
But I would recommend a dedicated network for interconnection and management
and not run that over the public/sip facing interfaces.

 

/b

 

On Mar 18, 2010, at 12:37 PM, Todd wrote:





Thanks folks!  I believe we will be doing most of our coding in php.  We are
also going to have independent DB servers.. I was also wondering if putting
the FS database on a DB server is a god idea, or if it should be left on the
FS box.  'Preciate it yall.  Learning is phun

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.791 / Virus Database: 271.1.1/2752 - Release Date: 03/18/10
03:33:00

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From infos at madovsky.org  Thu Mar 18 10:59:51 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 13:59:51 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <ACAA12B8CE2849E0977E3BE902C18B21@MOBILEE1705><191c3a031003171255i73032e3eo2cfa796d281fe16e@mail.gmail.com><14269D4B83DB472CB86C11B25730CD3F@MOBILEE1705>
	<191c3a031003171337x4ec01fa1i9a4df2a0d777c9ea@mail.gmail.com>
Message-ID: <74540B793B4E4E1B869CBD7D954C1383@MOBILEE1705>

Hi Anthony,

after 5 days of tests to resolve the hell of sendmail segmentation fault,
my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

    <param name="mailer-app" value="/usr/sbin/ssmtp"/>
    <param name="mailer-app-args" value="-t"/>

and apply the patch attached (no choice since it needs filename as argument and not STDIN).

hope it will help people who have sendmail segmentation fault problem.

Regards

Franck

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, March 17, 2010 4:37 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  FYI if you made the change in your working copy you could generate a patch with

  svn diff > patch.txt

  After looking at your change, the problem is that many people depend on this working the way it currently is.
  Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



  On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

    Sorry Anthony I never made an official update in C language with diff patch
    so the change is at the line 620, I commented out the original line

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 3:55 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      could you get a svn diff of the switch_utils.c so we can  see what the change was?




      On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

        Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
        it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

        I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

        Regards

        Franck
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Tuesday, March 16, 2010 4:35 PM
          Subject: Re: [Freeswitch-users] voicemail options


          ok, I tried 50 times to leave a message,
          it worked only one time.
          It seems that sendmail needs to be free to send the email....

            ----- Original Message ----- 
            From: devel at thom.fr.eu.org 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 3:59 PM
            Subject: Re: [Freeswitch-users] voicemail options


            Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

            I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



            Fran?ois





            De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
            Envoy? : mardi 16 mars 2010 20:30
            ? : freeswitch-users at lists.freeswitch.org
            Objet : Re: [Freeswitch-users] voicemail options



            are you running FS as root or a regular user?
            maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



            On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

            I noticed that sendmail makes a segfault when FS is trying to send email

            and sendmail has some emails locked in folder queue....



            Regards



            Franck


            _______________________________________________
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            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
            http://www.freeswitch.org




            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



--------------------------------------------------------------------


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            FreeSWITCH-users at lists.freeswitch.org
            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Thu Mar 18 11:03:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 11:03:30 -0700
Subject: [Freeswitch-users] Calls
In-Reply-To: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>
References: <d021d3be1003181046h2b7d1066m3ab5af6ed16e500@mail.gmail.com>
Message-ID: <87f2f3b91003181103j60a7f91eu5826b38f5b2cb400@mail.gmail.com>

show channels

On Thu, Mar 18, 2010 at 10:46 AM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello,
>
> Is it possible to list calls that are terminated in IVR's, are parked
> or in early state?
> "show calls" api just lists established calls, correct?
>
> Best regards,
>  Jonas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steve.d.ward at gmail.com  Thu Mar 18 11:04:56 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Thu, 18 Mar 2010 14:04:56 -0400
Subject: [Freeswitch-users] mod_fifo recording with fifo_record_template
Message-ID: <4ea6e8f21003181104j3775155y4b33ef27798beac6@mail.gmail.com>

I'm looking to use fifo_record_template to specify uniquely named recording
files.  Here is the situation:

I'm using a fifo with on-hook members.  I would like each fifo call to be
recorded, with each recording-file named uniquely.

Thanks to some helpful IRC talk, I have learned that the following works
well:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/my_recording.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

However, this records each and every fifo call to the file (in my case):
/usr/local/freeswitch/recordings/my_recording.wav

Every new call overwrites the file.

Instead of a statically defined file name, I would like a unique file name
for every fifo call.

I don't know how to make reference to channel variables in this definition.
I tried the following, but it results in an error:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/{${uuid}.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

The error I see on the CLI when a call goes into the FIFO is this:

2010-03-18 13:51:11.289448 [CRIT] switch_channel.c:759 Invalid data
(${fifo_record_template} contains a variable)

It was also suggested to me that I escape the '$' character on the
variable.  Like this:

<fifo name="Outside_Call@$${domain}" importance="0">
   <member timeout="20" simo="1"
lag="0">{fifo_record_template=$${base_dir}/recordings/{\${uuid}.wav,fifo_member_wait=nowait}user/35351@
$${domain}</member>
</fifo>

This also resulted in:

2010-03-18 13:52:37.014742 [CRIT] switch_channel.c:759 Invalid data
(${fifo_record_template} contains a variable)

The recording file doesn't get generated.

Any recommendations on using fifo_record_template to generate recording file
names that are unique to each call?  Is there a way to get channel variables
to be evaluated and populate the value of fifo_record_template?

Or does fifo_record_template only support static values (meaning I'd have to
do a record_session in dial plan?)?

Thanks for any feedback.
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From infos at madovsky.org  Thu Mar 18 11:10:55 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 14:10:55 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
Message-ID: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>

Also no need to uninstall sendmail, just install ssmtp and that's it !
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 1:59 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Hi Anthony,

  after 5 days of tests to resolve the hell of sendmail segmentation fault,
  my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
  I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
  So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

  configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

      <param name="mailer-app" value="/usr/sbin/ssmtp"/>
      <param name="mailer-app-args" value="-t"/>

  and apply the patch attached (no choice since it needs filename as argument and not STDIN).

  hope it will help people who have sendmail segmentation fault problem.

  Regards

  Franck

    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Wednesday, March 17, 2010 4:37 PM
    Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


    FYI if you made the change in your working copy you could generate a patch with

    svn diff > patch.txt

    After looking at your change, the problem is that many people depend on this working the way it currently is.
    Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



    On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

      Sorry Anthony I never made an official update in C language with diff patch
      so the change is at the line 620, I commented out the original line

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Wednesday, March 17, 2010 3:55 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        could you get a svn diff of the switch_utils.c so we can  see what the change was?




        On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

          Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
          it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

          I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

          Regards

          Franck
            ----- Original Message ----- 
            From: Madovsky 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Tuesday, March 16, 2010 4:35 PM
            Subject: Re: [Freeswitch-users] voicemail options


            ok, I tried 50 times to leave a message,
            it worked only one time.
            It seems that sendmail needs to be free to send the email....

              ----- Original Message ----- 
              From: devel at thom.fr.eu.org 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 16, 2010 3:59 PM
              Subject: Re: [Freeswitch-users] voicemail options


              Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

              I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



              Fran?ois





              De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
              Envoy? : mardi 16 mars 2010 20:30
              ? : freeswitch-users at lists.freeswitch.org
              Objet : Re: [Freeswitch-users] voicemail options



              are you running FS as root or a regular user?
              maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



              On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

              I noticed that sendmail makes a segfault when FS is trying to send email

              and sendmail has some emails locked in folder queue....



              Regards



              Franck


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              -- 
              Anthony Minessale II

              FreeSWITCH http://www.freeswitch.org/
              ClueCon http://www.cluecon.com/
              Twitter: http://twitter.com/FreeSWITCH_wire

              AIM: anthm
              MSN:anthony_minessale at hotmail.com
              GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
              IRC: irc.freenode.net #freeswitch

              FreeSWITCH Developer Conference
              sip:888 at conference.freeswitch.org
              iax:guest at conference.freeswitch.org/888
              googletalk:conf+888 at conference.freeswitch.org
              pstn:+19193869900



------------------------------------------------------------------


              _______________________________________________
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              FreeSWITCH-users at lists.freeswitch.org
              http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
              UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
              http://www.freeswitch.org


          _______________________________________________
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          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org





        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



------------------------------------------------------------------------


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900



----------------------------------------------------------------------------


    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From will.traenkle at yahoo.com  Thu Mar 18 11:15:36 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Thu, 18 Mar 2010 11:15:36 -0700 (PDT)
Subject: [Freeswitch-users] dial by name directory // mod_directory?
Message-ID: <487344.64278.qm@web57604.mail.re1.yahoo.com>

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?

I heard this was a feature that was implemented or going to be implemented in freeswitch.

If it has been implemented, would you send me info on how to configure it?

If it has not been implemented, would you state when it is planned to be implemented?

I believe the module name is mod_directory, is that correct?

Thanks,

-Will



      
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From xengelpublicx at gmail.com  Thu Mar 18 11:17:17 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Thu, 18 Mar 2010 21:17:17 +0300
Subject: [Freeswitch-users] intercept in fifo queue
In-Reply-To: <87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>
References: <eb0385731003151321ta75fa66ub09216edf3b3cdb4@mail.gmail.com>
	<87f2f3b91003161419q5d4fb8dbr754bad70238ffdea@mail.gmail.com>
	<eb0385731003170522n1d8229ces509d7a8cf09deed7@mail.gmail.com>
	<87f2f3b91003171211g717daf55h79dfcbcc2f7847c0@mail.gmail.com>
	<eb0385731003172311k1d4d3774g160c49e42ea626d2@mail.gmail.com>
	<87f2f3b91003172359s29714186g2e9c4c330900bcb4@mail.gmail.com>
	<eb0385731003180039pf0bd9bcr9bfade3c912c6941@mail.gmail.com>
	<87f2f3b91003180857i3a343f85x99d417cf18f81980@mail.gmail.com>
Message-ID: <eb0385731003181117o3140ec6bv325d14ccdd14654c@mail.gmail.com>

ok. I will try to realize, what I want with "agent access" extension.
Thanks for your help.

On Thu, Mar 18, 2010 at 6:57 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Thu, Mar 18, 2010 at 12:39 AM, Vladimir Elizarov
> <xengelpublicx at gmail.com> wrote:
>>
>> If she leaves the queue, the queue will not be agents. A user simply
>> intercepting the call should be able to transfer the call to your
>> phone when the Secretary is not there.
>
> I'm sorry but I don't understand. If the receptionist logs out of the queue
> then calls that are queued up will go to other agents (if available) or stay
> in queue until the receptionist gets back. Also, you can do the "agent
> access" extension I mentioned earlier so that non-agents can take callers
> out of the queue.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Best regards, Vladimir Elizarov



From msc at freeswitch.org  Thu Mar 18 11:28:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 11:28:44 -0700
Subject: [Freeswitch-users] dial by name directory // mod_directory?
In-Reply-To: <487344.64278.qm@web57604.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
Message-ID: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>

On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle
<will.traenkle at yahoo.com>wrote:

> Has freeswitch implemented the "Dial by Name" feature where a caller can
> enter the first x number of characters of an extensions first or last name
> and have the operator either connect the user to the result and/or state the
> result's extension number?
>
> I heard this was a feature that was implemented or going to be implemented
> in freeswitch.
>
> If it has been implemented, would you send me info on how to configure it?
>
> If it has not been implemented, would you state when it is planned to be
> implemented?
>
> I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report
back.
-MC
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From lists at redbonez.net  Thu Mar 18 11:41:19 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 12:41:19 -0600
Subject: [Freeswitch-users] Disable call waiting
Message-ID: <026101cac6ca$9b010830$d1031890$@net>

I apologize if this has been answered before, but I can't seem to find
anything on google. 

 

Is there a way to disable call waiting in FreeSWITCH? 

 

-Adam

 

 

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From brian at freeswitch.org  Thu Mar 18 11:46:06 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 13:46:06 -0500
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <026101cac6ca$9b010830$d1031890$@net>
References: <026101cac6ca$9b010830$d1031890$@net>
Message-ID: <DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>

FreeSWITCH is NOT involved in call waiting.  Thats an endpoint specific feature.

If you have an endpoint that doesn't let you see mod_limit.

/b

On Mar 18, 2010, at 1:41 PM, Adam Ford wrote:

> I apologize if this has been answered before, but I can?t seem to find anything on google.
>  
> Is there a way to disable call waiting in FreeSWITCH?
>  
> -Adam

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From lists at redbonez.net  Thu Mar 18 11:51:36 2010
From: lists at redbonez.net (Adam Ford)
Date: Thu, 18 Mar 2010 12:51:36 -0600
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>
References: <026101cac6ca$9b010830$d1031890$@net>
	<DD798C3F-D6EB-468A-9FC2-7FE2537D997B@freeswitch.org>
Message-ID: <027201cac6cc$0aa4c900$1fee5b00$@net>

That would explain the lack of info.  Thanks Brian

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, March 18, 2010 12:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Disable call waiting

 

FreeSWITCH is NOT involved in call waiting.  Thats an endpoint specific
feature.

 

If you have an endpoint that doesn't let you see mod_limit.

 

/b

 

On Mar 18, 2010, at 1:41 PM, Adam Ford wrote:





I apologize if this has been answered before, but I can't seem to find
anything on google.

 

Is there a way to disable call waiting in FreeSWITCH?

 

-Adam

 

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From anthony.minessale at gmail.com  Thu Mar 18 11:56:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 13:56:10 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1268900370911-4755257.post@n2.nabble.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268672644810-4738333.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
Message-ID: <191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>

didn't I ask you to type show channels between the calls once they were
established?
are these bypass media calls?


On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> You're right, sorry.
>
> I updated to trunk (rev 17031) and the problem is still here.
>
> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
> They
> contain comments and commands that I used.
>
> Scenario:
> 1st call is made and is parked. Then 2nd and 3rd calls are:
> 1) dialed
> 2) parked
> 3) bridged with 1st call
> 4) hanged up
>
> It is OK for now.
> Then a 4th call is dialed, parked, bridged and killed. After killing the
> 4th
> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>
>
> I hope this description and logs help resolve the issue.
> Please let me know if you need any more information.
>
> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From freeswitch-users-list at metik.com  Thu Mar 18 12:01:21 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Thu, 18 Mar 2010 15:01:21 -0400
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <026101cac6ca$9b010830$d1031890$@net>
References: <026101cac6ca$9b010830$d1031890$@net>
Message-ID: <4BA27881.3090205@metik.com>

I recommend addressing this issue via endpoint configuration. If you 
really must to do it, I have successfully used "Mod Limit 
(http://wiki.freeswitch.org/wiki/Mod_limit) to do it in an effort to 
band-aid a call waiting bug exhibited by a SIP-based DSLAM (until the 
manufacturer could fix it). Not desirable--but works.

-Metik


Adam Ford wrote:
>
> I apologize if this has been answered before, but I can?t seem to find 
> anything on google.
>
> Is there a way to disable call waiting in FreeSWITCH?
>
> -Adam
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From anthony.minessale at gmail.com  Thu Mar 18 12:03:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:03:54 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<191c3a031003151021l7de8a5d8jf699f05607ea6e7e@mail.gmail.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
Message-ID: <191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>

even better

also do

sofia profile internal siptrace on

to include the sip traffic


On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> didn't I ask you to type show channels between the calls once they were
> established?
> are these bypass media calls?
>
>
>
> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
> maciej.aniserowicz at gmail.com> wrote:
>
>>
>> You're right, sorry.
>>
>> I updated to trunk (rev 17031) and the problem is still here.
>>
>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>> They
>> contain comments and commands that I used.
>>
>> Scenario:
>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>> 1) dialed
>> 2) parked
>> 3) bridged with 1st call
>> 4) hanged up
>>
>> It is OK for now.
>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>> 4th
>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>
>>
>> I hope this description and logs help resolve the issue.
>> Please let me know if you need any more information.
>>
>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>> --
>> View this message in context:
>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 18 12:08:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:08:19 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
Message-ID: <191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>

Never mind,

I was right the first time 6 emails ago:  It's plenty obvious in this trace:

You bridged the 2 legs and you hungup on the A-leg when B-leg was not
answered yet.

here is your dialstring that tells us the problem.


bgapi originate {originate_timeout=20}[ignore_early_media=true,max_forwards=
999999,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
sofia/gateway/prospect_gateway/13007310757871097512282 &park()


ignore_early_media=true is not valid inside [] brackets only the global
section {} brackets.

if you move it there then you will never have any unanswered channels to
work with.



On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> even better
>
> also do
>
> sofia profile internal siptrace on
>
> to include the sip traffic
>
>
>
> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> didn't I ask you to type show channels between the calls once they were
>> established?
>> are these bypass media calls?
>>
>>
>>
>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>> maciej.aniserowicz at gmail.com> wrote:
>>
>>>
>>> You're right, sorry.
>>>
>>> I updated to trunk (rev 17031) and the problem is still here.
>>>
>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>> They
>>> contain comments and commands that I used.
>>>
>>> Scenario:
>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>> 1) dialed
>>> 2) parked
>>> 3) bridged with 1st call
>>> 4) hanged up
>>>
>>> It is OK for now.
>>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>>> 4th
>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>>
>>>
>>> I hope this description and logs help resolve the issue.
>>> Please let me know if you need any more information.
>>>
>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>>> --
>>> View this message in context:
>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Mar 18 12:16:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 14:16:49 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
Message-ID: <191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>

furthermore,

 to make sure this is over,

I added

uuid_bridge_continue_on_cancel=true chanvar you can set on A-leg to cause it

to continue in the dp even in your incorrect scenario


On Thu, Mar 18, 2010 at 2:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Never mind,
>
> I was right the first time 6 emails ago:  It's plenty obvious in this
> trace:
>
> You bridged the 2 legs and you hungup on the A-leg when B-leg was not
> answered yet.
>
> here is your dialstring that tells us the problem.
>
>
> bgapi originate {originate_timeout=20}[
> ignore_early_media=true,max_forwards=999999
> ,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
> sofia/gateway/prospect_gateway/13007310757871097512282 &park()
>
>
> ignore_early_media=true is not valid inside [] brackets only the global
> section {} brackets.
>
> if you move it there then you will never have any unanswered channels to
> work with.
>
>
>
> On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> even better
>>
>> also do
>>
>> sofia profile internal siptrace on
>>
>> to include the sip traffic
>>
>>
>>
>> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> didn't I ask you to type show channels between the calls once they were
>>> established?
>>> are these bypass media calls?
>>>
>>>
>>>
>>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>>> maciej.aniserowicz at gmail.com> wrote:
>>>
>>>>
>>>> You're right, sorry.
>>>>
>>>> I updated to trunk (rev 17031) and the problem is still here.
>>>>
>>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>>> They
>>>> contain comments and commands that I used.
>>>>
>>>> Scenario:
>>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>>> 1) dialed
>>>> 2) parked
>>>> 3) bridged with 1st call
>>>> 4) hanged up
>>>>
>>>> It is OK for now.
>>>> Then a 4th call is dialed, parked, bridged and killed. After killing the
>>>> 4th
>>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL cause.
>>>>
>>>>
>>>> I hope this description and logs help resolve the issue.
>>>> Please let me know if you need any more information.
>>>>
>>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit host.
>>>> --
>>>> View this message in context:
>>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tjardick at vanderkraan.net  Thu Mar 18 13:02:31 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 21:02:31 +0100
Subject: [Freeswitch-users] moh background in Lua
Message-ID: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>

Hi.

I'm trying to create a lua script but would like a certain task (music  
on hold) to be 'backgrounded':

-- Loop while the call is active
while (session:ready() == true) do
    -- do something here
    session:execute("playback", moh)
    freeswitch.consoleLog("INFO","Started music on hold.\n");
    -- do some stuff while playing music
    session:sleep(30000);
   -- Maybe we need a way to stop music or playback will stop  
automatically?
   session:execute("playback", "/sounds/we_will_answer_asap.wav")
   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
end

"ofcourse" currently the only thin that happens is the music and my  
loop does not continue.
Is there a way to start/stop music in the background?

I guess in Asterisk language i will need the command StartMusicOnHold...

Thanks for any suggestions.

Regards,

Tj



From anthony.minessale at gmail.com  Thu Mar 18 13:10:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 15:10:31 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
Message-ID: <191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>

The dilemma is we also need solution that is compat with windows, bsd, mac
etc.

That is sort of interesting that the pipe is breaking, it would be nice to
figure out why since not everybody
seems to have it.

I am still wondering if it's a stack size running out issue in the child
thread from system.
Are you running FS as root or as a non-privileged user, i'm wondering
because maybe if its a regular user, its not able to increase the stack size
up to 8megs

maybe you can push in strace into the command so it will show why it breaks.



On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

>  Also no need to uninstall sendmail, just install ssmtp and that's it !
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 18, 2010 1:59 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> Hi Anthony,
>
> after 5 days of tests to resolve the hell of sendmail segmentation fault,
> my conclusion is there is definitely a problem with the combination of cat,
> pipe and sendmail or other mail app.
> I made a php script that fits the actual switch_utils.c exec, but it's the
> same, it works one time on 10.
> So the robustest solution  is to replace sendmail by ssmtp (yum install
> ssmtp or apt-get ssmtp etc...)
>
> configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the
> switch.conf.xml like this
>
>     <param name="mailer-app" value="/usr/sbin/ssmtp"/>
>     <param name="mailer-app-args" value="-t"/>
>
> and apply the patch attached (no choice since it needs filename as argument
> and not STDIN).
>
>
> hope it will help people who have sendmail segmentation fault problem.
>
> Regards
>
> Franck
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Wednesday, March 17, 2010 4:37 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> FYI if you made the change in your working copy you could generate a patch
> with
>
> svn diff > patch.txt
>
> After looking at your change, the problem is that many people depend on
> this working the way it currently is.
> Would you consider making a script that mimics the current behavior of a
> mailer app and read the file from stdin and deliver the subsequent file so
> it is backwards-compatible with the way everything already works?
>
>
> On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Sorry Anthony I never made an official update in C language with diff
>> patch
>> so the change is at the line 620, I commented out the original line
>>
>>
>>  ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Wednesday, March 17, 2010 3:55 PM
>> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>>
>> could you get a svn diff of the switch_utils.c so we can  see what the
>> change was?
>>
>>
>>
>> On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  Finally I succeed to send voicemail with sendmail through a little php
>>> script (could be Perl also)
>>> it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug
>>> doesn't exist.
>>>
>>> I attached my php script and switch_utils.c. Not sure if the FS
>>> developers will accept my modification
>>>
>>> Regards
>>>
>>> Franck
>>>
>>> ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Tuesday, March 16, 2010 4:35 PM
>>> *Subject:* Re: [Freeswitch-users] voicemail options
>>>
>>> ok, I tried 50 times to leave a message,
>>> it worked only one time.
>>> It seems that sendmail needs to be free to send the email....
>>>
>>>
>>> ----- Original Message -----
>>> *From:* devel at thom.fr.eu.org
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Tuesday, March 16, 2010 3:59 PM
>>> *Subject:* Re: [Freeswitch-users] voicemail options
>>>
>>>  Are you running debian, I have the same problem on debian x64 (already
>>> mentioned here
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html)
>>> and cannot fix it.
>>>
>>> I don?t know if it?s a priviledge problem, as I got the same results
>>> running FS with or without ?u.
>>>
>>>
>>>
>>> Fran?ois
>>>
>>>
>>>
>>>
>>>
>>> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Anthony
>>> Minessale
>>> *Envoy? :* mardi 16 mars 2010 20:30
>>> *? :* freeswitch-users at lists.freeswitch.org
>>> *Objet :* Re: [Freeswitch-users] voicemail options
>>>
>>>
>>>
>>> are you running FS as root or a regular user?
>>> maybe there is a permission or resource limitation problem preventing it
>>> from getting the larger stack space necessary?
>>>
>>>  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> I noticed that sendmail makes a segfault when FS is trying to send email
>>>
>>> and sendmail has some emails locked in folder queue....
>>>
>>>
>>>
>>> Regards
>>>
>>>
>>>
>>> Franck
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 18 13:19:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 15:19:11 -0500
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
Message-ID: <191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>

The script is executing in the foreground and is being executed on your
channel.

You cannot tell the channel to do something else because it's already
running your script so
it can only run something else on top of it.

you have 2 choices:

1) use the socket app in async mode and external scripts with an ESL plugin.
2) instead of executing the script on the session launch a script in a sep
thread with the current uuid
    ad an argument then execute park and let the script control the channel
remotely

Even better reconsider what you want to do and avoid working against the
grain of how the existing architecture is flowing and you will have the best
results.



On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <
tjardick at vanderkraan.net> wrote:

> Hi.
>
> I'm trying to create a lua script but would like a certain task (music
> on hold) to be 'backgrounded':
>
> -- Loop while the call is active
> while (session:ready() == true) do
>    -- do something here
>    session:execute("playback", moh)
>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>    -- do some stuff while playing music
>    session:sleep(30000);
>   -- Maybe we need a way to stop music or playback will stop
> automatically?
>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
> end
>
> "ofcourse" currently the only thin that happens is the music and my
> loop does not continue.
> Is there a way to start/stop music in the background?
>
> I guess in Asterisk language i will need the command StartMusicOnHold...
>
> Thanks for any suggestions.
>
> Regards,
>
> Tj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Thu Mar 18 13:19:45 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 18 Mar 2010 22:19:45 +0200
Subject: [Freeswitch-users] Disable call waiting
In-Reply-To: <4BA27881.3090205@metik.com>
References: <026101cac6ca$9b010830$d1031890$@net> <4BA27881.3090205@metik.com>
Message-ID: <d361ca7b1003181319j2202c25aj7fbf62ef504104c8@mail.gmail.com>

We do this by checking the core DB for the user's status. We do so since we
have a logic to decide whether to allow call waiting, send busy or send to
voicemail. The user's prefference is set in a database.

                        regards, __Yehavi:

2010/3/18 Metik <freeswitch-users-list at metik.com>

> I recommend addressing this issue via endpoint configuration. If you
> really must to do it, I have successfully used "Mod Limit
> (http://wiki.freeswitch.org/wiki/Mod_limit) to do it in an effort to
> band-aid a call waiting bug exhibited by a SIP-based DSLAM (until the
> manufacturer could fix it). Not desirable--but works.
>
> -Metik
>
>
> Adam Ford wrote:
> >
> > I apologize if this has been answered before, but I can?t seem to find
> > anything on google.
> >
> > Is there a way to disable call waiting in FreeSWITCH?
> >
> > -Adam
> >
> > ------------------------------------------------------------------------
>  >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From m.sobkow at marketelsystems.com  Thu Mar 18 13:31:00 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 14:31:00 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
Message-ID: <4BA28D84.6040109@marketelsystems.com>

I know eavesdrop is an application, not an API, so you have to use it 
differently.

I have an operator dialing in on extension 5000 being bridged to a 
client, and I'm getting those lovely little echoes that confirm the call 
is bridged (both operator and client softphones are sharing my one 
headset and mic.)

In the dialplan, I parse out the extension when someone dials in to 
665000 (last 4 digits are the extension to monitor), look up the 
appropriate operator UUID in my mnesia/Erlang tables, receive the "call" 
callback from Freeswitch, and build a proplists replacement with the 
execute-app-name set to "eavesdrop" and execute-app-arg set to the 
operator UUID.  I issue a sendmsg() command to Freeswitch, and it 
returns ok.

However, what happens to the extension that tried to eavesdrop is it 
gets two events: CHANNEL_PARK and SEND_MESSAGE.  The person making the 
eavesdrop call can't actually hear anything.

I've attached the Erlang callback module that the dialplan executes.  
The dialplan fragment is as follows:

<extension name=\"eavesdrop\">
    <condition field=\"destination_number\" 
expression=\"^66(\\d\\d\\d\\d)$\">
        <action application=\"answer\"/>
        <action application=\"set\" data=\"eavesdrop_extension=$1\" />
        <action application=\"erlang\" data=\"pbx_eavesdrop 
pursuit at testsrv\" />
    </condition>
</extension>

Unfortunately I can't think of any way to look up the UUID other than 
using the mnesia tables from the application.  When I tried the sample 
dialplans from Freeswitch, the eavesdrop module itself _did_ work.  I'm 
just having trouble cobbling together the "glue" for Erlang.

P.S. This is the last Freeswitch feature I have to code for this 
release, other than niggling cleanups like adding a few more voice 
prompts, which is easy enough as I've already sample code on how to do 
it.  With a little luck we'll make our April 1st ready-for-testing 
deadline!  We wouldn't have made it without Andrew's hints, tips, and 
tricks for using Erlang with Freeswitch. Thanks a lot! :D

-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 

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From fraserredmond at gmail.com  Thu Mar 18 13:45:19 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Thu, 18 Mar 2010 20:45:19 +0000
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl> 
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com> 
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com> 
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>
	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl> 
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
Message-ID: <c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>

Is there any advantage at all to putting the core database on your own
external db? Or is it just a control freak thing? I was looking at this the
other day, and only ever read that it was possible, not anything about why
you'd want to do it.

Cheers,
Fraser




On Thu, Mar 18, 2010 at 5:41 PM, Brian West <brian at freeswitch.org> wrote:

>  Its really up to you it works both ways.  But I would recommend a
> dedicated network for interconnection and management and not run that over
> the public/sip facing interfaces.
>
> /b
>
> On Mar 18, 2010, at 12:37 PM, Todd wrote:
>
> Thanks folks!  I believe we will be doing most of our coding in php.  We
> are also going to have independent DB servers?. I was also wondering if
> putting the FS database on a DB server is a god idea, or if it should be
> left on the FS box.  ?Preciate it yall.  Learning is phun
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tjardick at vanderkraan.net  Thu Mar 18 14:02:52 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 18 Mar 2010 22:02:52 +0100
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
	<191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
Message-ID: <D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>

Hi Anthony,

Thanks for the quick and detailed reply.

Could i pick your brain on the last remark, what i'd like to do is:

- play a configurable time of hold music (30 seconds in the example)  
but this will be a variable later
- play a message (and optionally detect dtmf if needed)
- if no dtmf then continue music

and loop the above.

So basically creating an inbound holding point for people to wait  
before their call is answered, but give them the option to leave the  
'queue' to direct them else where.

Would in the async ESL i be able to stop the stream of music or would  
i need to do a bridge to local/9999 and unbridge to continue when i  
want to? Would ESL be the best way to
do this ?

Thanks for the help so far.

Regards,

Tj


On 18 Mar 2010, at 21:19, Anthony Minessale wrote:

> The script is executing in the foreground and is being executed on  
> your channel.
>
> You cannot tell the channel to do something else because it's  
> already running your script so
> it can only run something else on top of it.
>
> you have 2 choices:
>
> 1) use the socket app in async mode and external scripts with an ESL  
> plugin.
> 2) instead of executing the script on the session launch a script in  
> a sep thread with the current uuid
>     ad an argument then execute park and let the script control the  
> channel remotely
>
> Even better reconsider what you want to do and avoid working against  
> the grain of how the existing architecture is flowing and you will  
> have the best results.
>
>
>
> On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <tjardick at vanderkraan.net 
> > wrote:
> Hi.
>
> I'm trying to create a lua script but would like a certain task (music
> on hold) to be 'backgrounded':
>
> -- Loop while the call is active
> while (session:ready() == true) do
>    -- do something here
>    session:execute("playback", moh)
>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>    -- do some stuff while playing music
>    session:sleep(30000);
>   -- Maybe we need a way to stop music or playback will stop
> automatically?
>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
> end
>
> "ofcourse" currently the only thin that happens is the music and my
> loop does not continue.
> Is there a way to start/stop music in the background?
>
> I guess in Asterisk language i will need the command  
> StartMusicOnHold...
>
> Thanks for any suggestions.
>
> Regards,
>
> Tj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From andrew at hijacked.us  Thu Mar 18 14:11:39 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Mar 2010 17:11:39 -0400
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <4BA28D84.6040109@marketelsystems.com>
References: <4BA28D84.6040109@marketelsystems.com>
Message-ID: <20100318211138.GF28551@hijacked.us>

I do it like this:

freeswitch:bgapi(Fnode, originate, "user/agentname" ++ " &eavesdrop(" ++ UUID ++ ")")

But user/agentname can be any FS dialstring (SIP/... or whatever). It
dials the endpoint and then the b leg is the eavesdrop() call.

Andrew



From msc at freeswitch.org  Thu Mar 18 14:12:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Mar 2010 14:12:32 -0700
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>
References: <1268643396699-4735736.post@n2.nabble.com>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<191c3a031003181216x30ae6bc9nfc9c0ea456274c8d@mail.gmail.com>
Message-ID: <87f2f3b91003181412j2859edb1mcd4a1939a57c624b@mail.gmail.com>

Doc'd here:
http://wiki.freeswitch.org/wiki/Variable_uuid_bridge_continue_on_cancel
-MC

On Thu, Mar 18, 2010 at 12:16 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> furthermore,
>
>  to make sure this is over,
>
> I added
>
> uuid_bridge_continue_on_cancel=true chanvar you can set on A-leg to cause
> it
> to continue in the dp even in your incorrect scenario
>
>
> On Thu, Mar 18, 2010 at 2:08 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Never mind,
>>
>> I was right the first time 6 emails ago:  It's plenty obvious in this
>> trace:
>>
>> You bridged the 2 legs and you hungup on the A-leg when B-leg was not
>> answered yet.
>>
>> here is your dialstring that tells us the problem.
>>
>>
>> bgapi originate {originate_timeout=20}[
>> ignore_early_media=true,max_forwards=999999
>> ,origination_uuid=9b825981-06cc-41ce-baf6-3dfaff2aba64]
>> sofia/gateway/prospect_gateway/13007310757871097512282 &park()
>>
>>
>> ignore_early_media=true is not valid inside [] brackets only the global
>> section {} brackets.
>>
>> if you move it there then you will never have any unanswered channels to
>> work with.
>>
>>
>>
>> On Thu, Mar 18, 2010 at 2:03 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> even better
>>>
>>> also do
>>>
>>> sofia profile internal siptrace on
>>>
>>> to include the sip traffic
>>>
>>>
>>>
>>> On Thu, Mar 18, 2010 at 1:56 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> didn't I ask you to type show channels between the calls once they were
>>>> established?
>>>> are these bypass media calls?
>>>>
>>>>
>>>>
>>>> On Thu, Mar 18, 2010 at 3:19 AM, Maciej Aniserowicz <
>>>> maciej.aniserowicz at gmail.com> wrote:
>>>>
>>>>>
>>>>> You're right, sorry.
>>>>>
>>>>> I updated to trunk (rev 17031) and the problem is still here.
>>>>>
>>>>> Logs produced by newest version: http://pastebin.freeswitch.org/12474.
>>>>> They
>>>>> contain comments and commands that I used.
>>>>>
>>>>> Scenario:
>>>>> 1st call is made and is parked. Then 2nd and 3rd calls are:
>>>>> 1) dialed
>>>>> 2) parked
>>>>> 3) bridged with 1st call
>>>>> 4) hanged up
>>>>>
>>>>> It is OK for now.
>>>>> Then a 4th call is dialed, parked, bridged and killed. After killing
>>>>> the 4th
>>>>> call, the 1st one is immediately hanged up with ORIGINATOR_CANCEL
>>>>> cause.
>>>>>
>>>>>
>>>>> I hope this description and logs help resolve the issue.
>>>>> Please let me know if you need any more information.
>>>>>
>>>>> FS is started on Win2003 32 bit run in VirtualPC on Windows7 64bit
>>>>> host.
>>>>> --
>>>>> View this message in context:
>>>>> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4755257.html
>>>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:+19193869900
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Mar 18 14:20:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Mar 2010 16:20:40 -0500
Subject: [Freeswitch-users] moh background in Lua
In-Reply-To: <D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>
References: <95E38317-4796-4444-8670-B450FD0C8231@vanderkraan.net>
	<191c3a031003181319y74a3320cgca99917986209f3e@mail.gmail.com>
	<D31C176D-E00F-4BAD-BE0F-28C35F3F2FEA@vanderkraan.net>
Message-ID: <191c3a031003181420n267d32f0oce0c3282c17e3385@mail.gmail.com>

You could use mod_fifo with the chimelist configured and avoid trying to
replicate this manually.

you can certainly do it from ESL because you have a socket connection to the
channel and all the commands you send will be in the background.


On Thu, Mar 18, 2010 at 4:02 PM, Tjardick van der Kraan <
tjardick at vanderkraan.net> wrote:

> Hi Anthony,
>
> Thanks for the quick and detailed reply.
>
> Could i pick your brain on the last remark, what i'd like to do is:
>
> - play a configurable time of hold music (30 seconds in the example) but
> this will be a variable later
> - play a message (and optionally detect dtmf if needed)
> - if no dtmf then continue music
>
> and loop the above.
>
> So basically creating an inbound holding point for people to wait before
> their call is answered, but give them the option to leave the 'queue' to
> direct them else where.
>
> Would in the async ESL i be able to stop the stream of music or would i
> need to do a bridge to local/9999 and unbridge to continue when i want to?
> Would ESL be the best way to
> do this ?
>
> Thanks for the help so far.
>
> Regards,
>
> Tj
>
>
> On 18 Mar 2010, at 21:19, Anthony Minessale wrote:
>
> The script is executing in the foreground and is being executed on your
> channel.
>
> You cannot tell the channel to do something else because it's already
> running your script so
> it can only run something else on top of it.
>
> you have 2 choices:
>
> 1) use the socket app in async mode and external scripts with an ESL
> plugin.
> 2) instead of executing the script on the session launch a script in a sep
> thread with the current uuid
>     ad an argument then execute park and let the script control the channel
> remotely
>
> Even better reconsider what you want to do and avoid working against the
> grain of how the existing architecture is flowing and you will have the best
> results.
>
>
>
> On Thu, Mar 18, 2010 at 3:02 PM, Tjardick van der Kraan <
> tjardick at vanderkraan.net> wrote:
>
>> Hi.
>>
>> I'm trying to create a lua script but would like a certain task (music
>> on hold) to be 'backgrounded':
>>
>> -- Loop while the call is active
>> while (session:ready() == true) do
>>    -- do something here
>>    session:execute("playback", moh)
>>    freeswitch.consoleLog("INFO","Started music on hold.\n");
>>    -- do some stuff while playing music
>>    session:sleep(30000);
>>   -- Maybe we need a way to stop music or playback will stop
>> automatically?
>>   session:execute("playback", "/sounds/we_will_answer_asap.wav")
>>   freeswitch.consoleLog("INFO","Play'ed announcement.\n");
>> end
>>
>> "ofcourse" currently the only thin that happens is the music and my
>> loop does not continue.
>> Is there a way to start/stop music in the background?
>>
>> I guess in Asterisk language i will need the command StartMusicOnHold...
>>
>> Thanks for any suggestions.
>>
>> Regards,
>>
>> Tj
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Thu Mar 18 14:24:41 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 17:24:41 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
Message-ID: <3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>

Ok, I read that there are some guys who had the same problem here.
I run FS as root, but sendmail as chroot smmsp user. You're right maybe
it's a thread child problem. I saw there is a windows condition already in your code ?
and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
I will experiment with strace tonight or tomorrow
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 4:10 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  The dilemma is we also need solution that is compat with windows, bsd, mac etc.

  That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
  seems to have it.

  I am still wondering if it's a stack size running out issue in the child thread from system.
  Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

  maybe you can push in strace into the command so it will show why it breaks.




  On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

    Also no need to uninstall sendmail, just install ssmtp and that's it !
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, March 18, 2010 1:59 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      Hi Anthony,

      after 5 days of tests to resolve the hell of sendmail segmentation fault,
      my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
      I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
      So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

      configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

          <param name="mailer-app" value="/usr/sbin/ssmtp"/>
          <param name="mailer-app-args" value="-t"/>

      and apply the patch attached (no choice since it needs filename as argument and not STDIN).
       
    hope it will help people who have sendmail segmentation fault problem.

    Regards

    Franck

      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Wednesday, March 17, 2010 4:37 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      FYI if you made the change in your working copy you could generate a patch with

      svn diff > patch.txt

      After looking at your change, the problem is that many people depend on this working the way it currently is.
      Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



      On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

        Sorry Anthony I never made an official update in C language with diff patch
        so the change is at the line 620, I commented out the original line

          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Wednesday, March 17, 2010 3:55 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


          could you get a svn diff of the switch_utils.c so we can  see what the change was?




          On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

            Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
            it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

            I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

            Regards

            Franck
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Tuesday, March 16, 2010 4:35 PM
              Subject: Re: [Freeswitch-users] voicemail options


              ok, I tried 50 times to leave a message,
              it worked only one time.
              It seems that sendmail needs to be free to send the email....

                ----- Original Message ----- 
                From: devel at thom.fr.eu.org 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 16, 2010 3:59 PM
                Subject: Re: [Freeswitch-users] voicemail options


                Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                Fran?ois





                De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                Envoy? : mardi 16 mars 2010 20:30
                ? : freeswitch-users at lists.freeswitch.org
                Objet : Re: [Freeswitch-users] voicemail options



                are you running FS as root or a regular user?
                maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                I noticed that sendmail makes a segfault when FS is trying to send email

                and sendmail has some emails locked in folder queue....



                Regards



                Franck


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                -- 
                Anthony Minessale II

                FreeSWITCH http://www.freeswitch.org/
                ClueCon http://www.cluecon.com/
                Twitter: http://twitter.com/FreeSWITCH_wire

                AIM: anthm
                MSN:anthony_minessale at hotmail.com
                GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
                IRC: irc.freenode.net #freeswitch

                FreeSWITCH Developer Conference
                sip:888 at conference.freeswitch.org
                iax:guest at conference.freeswitch.org/888
                googletalk:conf+888 at conference.freeswitch.org
                pstn:+19193869900



----------------------------------------------------------------


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            http://www.freeswitch.org





          -- 
          Anthony Minessale II

          FreeSWITCH http://www.freeswitch.org/
          ClueCon http://www.cluecon.com/
          Twitter: http://twitter.com/FreeSWITCH_wire

          AIM: anthm
          MSN:anthony_minessale at hotmail.com
          GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
          IRC: irc.freenode.net #freeswitch

          FreeSWITCH Developer Conference
          sip:888 at conference.freeswitch.org
          iax:guest at conference.freeswitch.org/888
          googletalk:conf+888 at conference.freeswitch.org
          pstn:+19193869900



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      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


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  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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From m.sobkow at marketelsystems.com  Thu Mar 18 14:36:53 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 15:36:53 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <20100318211138.GF28551@hijacked.us>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
Message-ID: <4BA29CF5.1050101@marketelsystems.com>

Ok, so that new originate command will give me another UUID, and I'll 
have to bridge it to the original call's UUID, correct?  i.e. Someone 
dials the 66xxxx extension, it calls back to Erlang and parks.  Erlang 
issues the originate command, returns a UUID for the new call, and I do 
a uuid_bridge so the caller can hear the eavesdrop.

The only thing I'm a little puzzled by is the user/agentname string.  
The caller is already on the phone, so how can Freeswitch call them?

Alternatively maybe I could set the operator UUID into a variable and 
establish the call through a dialplan extension, but I think that ends 
up with the same problem of the caller being busy.

I do have one other option -- we have a softphone built into our 
front-end application, so I could have a button press/menu selection on 
that application issue a command to Erlang, which would mean it could 
call the softphone because it's not busy trying to dial a special 66xxxx 
extension.  That actually might be the best solution, because then there 
is no "secret" extension in the dialplan that unauthorized users might 
learn about.  I just need to have one of our other developers provide an 
API to retrieve the configured extension/registration information for 
the softphone so it can pass along the extension to Erlang for building 
a dial string.  The only "downside" to that approach is that it 
_requires_ our client application to work, but I think our user 
community will be ok with that because managers need that application 
for a whole host of features, not just the softphone.

Andrew Thompson wrote:
> I do it like this:
>
> freeswitch:bgapi(Fnode, originate, "user/agentname" ++ " &eavesdrop(" ++ UUID ++ ")")
>
> But user/agentname can be any FS dialstring (SIP/... or whatever). It
> dials the endpoint and then the b leg is the eavesdrop() call.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
Mark Sobkow
Senior Developer
MarkeTel Multi-Line Dialing Systems LTD.
428 Victoria Ave
Regina, SK S4N-0P6
Toll-Free: 800-289-8616-X533
Local: 306-359-6893-X533
Fax: 306-359-6879
Email: m.sobkow at marketelsystems.com
Web: http://www.marketelsystems.com 




From lloydie.t at googlemail.com  Thu Mar 18 14:45:02 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Thu, 18 Mar 2010 21:45:02 +0000
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
Message-ID: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>

I eventually managed to get mod_xml_curl working for me. Now I trying to be
a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I
get an error when a phone tries to register.

2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
203 at 192.168.101.102]
You must define a domain called '192.168.101.102' in your directory and add
a user with the id="203" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
(REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
192.168.91.28

I think I am nearly there,but I suspect there is something wrong with the
resulting 'user directory' xml.
---------------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="User Directory">
<domain name="default">
<groups>
  <group name="default">
  <users>
  <user id="203">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      <param name="vm-password" value="1234"/>
      <param name="vm-disk-quota" value="30"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
      <variable name="accountcode" value="203"/>
      <variable name="day_bar" value="3"/>
      <variable name="night_bar" value="3"/>
      <variable name="effective_caller_id_name" value="Extension 203"/>
      <variable name="effective_caller_id_number" value="203"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_number}"/>
    </variables>
  </user>
  </users>
 </group>
</groups>
</domain>
</section>
</document>
--------------------------------------------------------------------

I think the Company Directory XML is OK
--------------------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="Company Directory">
<domain name="default">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="default"/>
  </variables>
</domain>
<domain name="phisystems.dom">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="phisystems.dom"/>
  </variables>
</domain>
</section>
</document>
--------------------------------------------------------------

I get the following result when I 'sofia status'
                     Name          Type
Data     State
=================================================================================================
                 internal       profile   sip:mod_sofia at 192.168.101.102:5060
RUNNING (0)
           phisystems.dom         alias
internal     ALIASED
            internal-ipv6       profile           sip:mod_sofia@[::1]:5060
RUNNING (0)
                  default         alias
internal     ALIASED
                 external       profile   sip:mod_sofia at 192.168.101.102:5080
RUNNING (0)
=================================================================================================

Thanks in advance
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From andrew at hijacked.us  Thu Mar 18 14:48:41 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Thu, 18 Mar 2010 17:48:41 -0400
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <4BA29CF5.1050101@marketelsystems.com>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
	<4BA29CF5.1050101@marketelsystems.com>
Message-ID: <20100318214841.GG28551@hijacked.us>

On Thu, Mar 18, 2010 at 03:36:53PM -0600, Mark Sobkow wrote:
> Ok, so that new originate command will give me another UUID, and I'll 
> have to bridge it to the original call's UUID, correct?  i.e. Someone 
> dials the 66xxxx extension, it calls back to Erlang and parks.  Erlang 
> issues the originate command, returns a UUID for the new call, and I do 
> a uuid_bridge so the caller can hear the eavesdrop.
> 
> The only thing I'm a little puzzled by is the user/agentname string.  
> The caller is already on the phone, so how can Freeswitch call them?
> 
> Alternatively maybe I could set the operator UUID into a variable and 
> establish the call through a dialplan extension, but I think that ends 
> up with the same problem of the caller being busy.
> 
> I do have one other option -- we have a softphone built into our 
> front-end application, so I could have a button press/menu selection on 
> that application issue a command to Erlang, which would mean it could 
> call the softphone because it's not busy trying to dial a special 66xxxx 
> extension.  That actually might be the best solution, because then there 
> is no "secret" extension in the dialplan that unauthorized users might 
> learn about.  I just need to have one of our other developers provide an 
> API to retrieve the configured extension/registration information for 
> the softphone so it can pass along the extension to Erlang for building 
> a dial string.  The only "downside" to that approach is that it 
> _requires_ our client application to work, but I think our user 
> community will be ok with that because managers need that application 
> for a whole host of features, not just the softphone.
>
Oh, I see what you wanted. Why not just bridge the call to an inline
dialplan that calls eavesdrop() then? That way you don't have to serve
up an extension or anything annoying.

something like

uuid_transfer CallerUUID 'eavesdrop:EavesdropUUID' inline

?

Andrew



From kenfulmer at icstechnologysolutions.com  Thu Mar 18 15:05:29 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 18 Mar 2010 17:05:29 -0500
Subject: [Freeswitch-users] PaeTec
Message-ID: <012001cac6e7$1f711490$5e533db0$@com>

Has anyone created a gateway to connect to PaeTec? 

 

Thanks,

 

Ken Fulmer

 

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From pavera at gmail.com  Wed Mar 17 22:07:05 2010
From: pavera at gmail.com (Tom Christensen)
Date: Wed, 17 Mar 2010 23:07:05 -0600
Subject: [Freeswitch-users] openzap no audio?
Message-ID: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>

I've got freeswitch installed with a digium te110p hooked up to a PRI.

calls through the system work fine SIP->SIP, calling and receiving calls
over the PRI works, (IE the signalling part, a call can be made and incoming
calls come in over the PRI and I can route them based on DID), however,
there is no audio to either party.  I've checked, and there is an RTP stream
running while the call is up, there is no reason it would be blocked (IE,
SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
few posts that talked about no audio for a few seconds after a call is
connected, or that calls would connect and then drop after a few seconds.
 This is different, I can leave the call up for many minutes, it doesn't
drop, and audio never starts coming through.

I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
the latest release of zaptel first, freeswitch wouldn't even load with that.

I'm going to try libpri now I think as I've exhausted all the other options
I can think of.

Any input/ideas would be greatly appreciated.

-Tom
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From pavera at gmail.com  Thu Mar 18 13:28:02 2010
From: pavera at gmail.com (Tom Christensen)
Date: Thu, 18 Mar 2010 14:28:02 -0600
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
Message-ID: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>

In my quest to replace an unstable and deadlock prone asterisk solution I
was able to get calls working over my PRI last night using freeswitch,
native openzap, and dahdi 2.2.1.  However, in order to make it work, I had
to disable all echo cancellation in dahdi.  If any of the 4 echo
cancellation algo's were specified in /etc/dahdi/system.conf then there was
no audio at all on the calls.  Signaling worked, and calls could be
made/received, but no audio in either direction.  As soon as I disabled EC,
audio worked (although, there was a good amount of echo).

Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very
poor voice quality (robotic sounding, distortions, dropping in and out).

This experience leads me to believe that possibly openzap isn't ready for
prime time?  Would I have been better served to include libpri in the
technology stack?

At this stage my "solution" looks like leaving asterisk where it is, and
using it as a media gateway, passing calls off to freeswitch via SIP or IAX
and using freeswitch as the app/registration/feature server.  I don't know
if that will necessarily be better, but I've had good luck with asterisk and
call quality tuning (the PRI here has always had pretty bad echo, but I've
been able to get asterisk EC working quite well on it), and it seems like
asterisk has most of its issues with stability in
transferring/parking/hold/voicemail/music on hold...

I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap using
a digium te110p.  The PRI is an NI2 t1 if that matters...

Any ideas or recommendations of things to try would be greatly appreciated.
I'd really like to avoid the dual server situation if at all possible.

Thanks,
-Tom
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From m.sobkow at marketelsystems.com  Thu Mar 18 15:23:01 2010
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Thu, 18 Mar 2010 16:23:01 -0600
Subject: [Freeswitch-users] How do I use eavesdrop?
In-Reply-To: <20100318214841.GG28551@hijacked.us>
References: <4BA28D84.6040109@marketelsystems.com>
	<20100318211138.GF28551@hijacked.us>
	<4BA29CF5.1050101@marketelsystems.com>
	<20100318214841.GG28551@hijacked.us>
Message-ID: <4BA2A7C5.4030509@marketelsystems.com>

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From Russell.Mosemann at cune.org  Thu Mar 18 17:48:46 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Thu, 18 Mar 2010 19:48:46 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
References: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
Message-ID: <AD5B87549E604DD3BBD73065A1EF6317@cune.pri>

Tom Christensen wrote:

> If any of the 4 echo
> cancellation algo's were specified in /etc/dahdi/system.conf then there
> was no audio at all on the calls.

I'm not glad this happened to you, but I'm glad to hear about the problem. I have the same problem, and I thought it was something specific to my setup. I even compiled OSLEC separately with the same results. The call completes, but there is absolutely no audio. It is completely silent. I suppose that is a pretty effective echo canceller. :-) I did not notice any errors in the console. If EC is removed, the audio works.

> Also, I had intermittent (like 1 out of 2 or 1 out of 3) calls with very
> poor voice quality (robotic sounding, distortions, dropping in and out).

I have not experienced that. There seems to be a brief audio dropout every couple of seconds during a call, like a couple of packets are being dropped every now and then, but I haven't dug into it, yet.

> This experience leads me to believe that possibly openzap isn't ready for
> prime time?  Would I have been better served to include libpri in the
> technology stack?

I'm sorry to report that I am using libpri, but that's interesting. I'm curious about the source of the problem and why EC would be affected. I don't see a connection.

> I'm running a current FS SVN from 3 days ago, dahdi 2.2.1, and openzap
> using a digium te110p.  The PRI is an NI2 t1 if that matters...

Exact same setup here. Hmm.

> Any ideas or recommendations of things to try would be greatly
> appreciated. I'd really like to avoid the dual server situation if at all
> possible.

Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all that bad here, but it would be nice to use EC.

----
Russell Mosemann




From gkuri at ieee.org  Thu Mar 18 19:48:15 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Mar 2010 19:48:15 -0700
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
Message-ID: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>

Hey guys, is the Caller ID data (number and name) supposed to be
passed between the phones when using SLA/SCA ? If you put a call on
hold on phone 1, and pick up the call on phone 2, number/name do not
show up. Likewise, if you go back to the same call on phone 1, the
name and number are not present. Is this normal? Do I smell a bounty
;) ?

Cheers,
Gabe



From infos at madovsky.org  Thu Mar 18 20:07:14 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 18 Mar 2010 23:07:14 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
Message-ID: <581ED17E563D47C5AD4E91CBF91DC727@MOBILEE1705>

but I think that the patch can also work with sendmail
as default config too... it needs a test with someone who uses
sendmail without seg fault...
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 18, 2010 5:24 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Ok, I read that there are some guys who had the same problem here.
  I run FS as root, but sendmail as chroot smmsp user. You're right maybe
  it's a thread child problem. I saw there is a windows condition already in your code ?
  and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
  I will experiment with strace tonight or tomorrow
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, March 18, 2010 4:10 PM
    Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


    The dilemma is we also need solution that is compat with windows, bsd, mac etc.

    That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
    seems to have it.

    I am still wondering if it's a stack size running out issue in the child thread from system.
    Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

    maybe you can push in strace into the command so it will show why it breaks.




    On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

      Also no need to uninstall sendmail, just install ssmtp and that's it !
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 18, 2010 1:59 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        Hi Anthony,

        after 5 days of tests to resolve the hell of sendmail segmentation fault,
        my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
        I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
        So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

        configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

            <param name="mailer-app" value="/usr/sbin/ssmtp"/>
            <param name="mailer-app-args" value="-t"/>

        and apply the patch attached (no choice since it needs filename as argument and not STDIN).
         
      hope it will help people who have sendmail segmentation fault problem.

      Regards

      Franck

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Wednesday, March 17, 2010 4:37 PM
        Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


        FYI if you made the change in your working copy you could generate a patch with

        svn diff > patch.txt

        After looking at your change, the problem is that many people depend on this working the way it currently is.
        Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



        On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

          Sorry Anthony I never made an official update in C language with diff patch
          so the change is at the line 620, I commented out the original line

            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Wednesday, March 17, 2010 3:55 PM
            Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


            could you get a svn diff of the switch_utils.c so we can  see what the change was?




            On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

              Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
              it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

              I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

              Regards

              Franck
                ----- Original Message ----- 
                From: Madovsky 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Tuesday, March 16, 2010 4:35 PM
                Subject: Re: [Freeswitch-users] voicemail options


                ok, I tried 50 times to leave a message,
                it worked only one time.
                It seems that sendmail needs to be free to send the email....

                  ----- Original Message ----- 
                  From: devel at thom.fr.eu.org 
                  To: freeswitch-users at lists.freeswitch.org 
                  Sent: Tuesday, March 16, 2010 3:59 PM
                  Subject: Re: [Freeswitch-users] voicemail options


                  Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                  I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                  Fran?ois





                  De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                  Envoy? : mardi 16 mars 2010 20:30
                  ? : freeswitch-users at lists.freeswitch.org
                  Objet : Re: [Freeswitch-users] voicemail options



                  are you running FS as root or a regular user?
                  maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                  On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                  I noticed that sendmail makes a segfault when FS is trying to send email

                  and sendmail has some emails locked in folder queue....



                  Regards



                  Franck


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                  -- 
                  Anthony Minessale II

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                  AIM: anthm
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                  pstn:+19193869900



--------------------------------------------------------------


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            -- 
            Anthony Minessale II

            FreeSWITCH http://www.freeswitch.org/
            ClueCon http://www.cluecon.com/
            Twitter: http://twitter.com/FreeSWITCH_wire

            AIM: anthm
            MSN:anthony_minessale at hotmail.com
            GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
            IRC: irc.freenode.net #freeswitch

            FreeSWITCH Developer Conference
            sip:888 at conference.freeswitch.org
            iax:guest at conference.freeswitch.org/888
            googletalk:conf+888 at conference.freeswitch.org
            pstn:+19193869900



--------------------------------------------------------------------


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            http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org





        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



------------------------------------------------------------------------


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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900



----------------------------------------------------------------------------


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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Thu Mar 18 20:23:44 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 18 Mar 2010 22:23:44 -0500
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
In-Reply-To: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
References: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
Message-ID: <6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>

I would post a bounty...   ;)

/b

On Mar 18, 2010, at 9:48 PM, Gabriel Kuri wrote:

> Hey guys, is the Caller ID data (number and name) supposed to be
> passed between the phones when using SLA/SCA ? If you put a call on
> hold on phone 1, and pick up the call on phone 2, number/name do not
> show up. Likewise, if you go back to the same call on phone 1, the
> name and number are not present. Is this normal? Do I smell a bounty
> ;) ?




From gkuri at ieee.org  Thu Mar 18 21:10:13 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 18 Mar 2010 21:10:13 -0700
Subject: [Freeswitch-users] Caller ID w/ SLA/SCA
In-Reply-To: <6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>
References: <8b1c9cda1003181948r5fb68ef6p5f059c4450026e25@mail.gmail.com>
	<6B21AF3F-CA4E-462D-90D7-18711D0D482D@freeswitch.org>
Message-ID: <8b1c9cda1003182110v411a7a7k5ccd316cf2f3e140@mail.gmail.com>

About how much, what kind of work is required for something like this?

Cheers,
Gabe

On Thu, Mar 18, 2010 at 8:23 PM, Brian West <brian at freeswitch.org> wrote:
> I would post a bounty... ? ;)
>
> /b
>
> On Mar 18, 2010, at 9:48 PM, Gabriel Kuri wrote:
>
>> Hey guys, is the Caller ID data (number and name) supposed to be
>> passed between the phones when using SLA/SCA ? If you put a call on
>> hold on phone 1, and pick up the call on phone 2, number/name do not
>> show up. Likewise, if you go back to the same call on phone 1, the
>> name and number are not present. Is this normal? Do I smell a bounty
>> ;) ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Thu Mar 18 21:32:29 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 19 Mar 2010 11:32:29 +0700
Subject: [Freeswitch-users] Indian and mod_say
Message-ID: <8ccbff061003182132x697f6539jf7b5d6934c0924a9@mail.gmail.com>

Can someone guide me about Indian language? i want to create mod_say_in

BG

Dome C.



From infos at madovsky.org  Thu Mar 18 21:55:40 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 00:55:40 -0400
Subject: [Freeswitch-users] variables in gateway profile
Message-ID: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>

Is it possible to create <variable> in gateway profiles like users profiles ?

Thanks

Franck
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From will.traenkle at yahoo.com  Fri Mar 19 00:42:08 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Mar 2010 00:42:08 -0700 (PDT)
Subject: [Freeswitch-users] dial by name directory // mod_directory?
In-Reply-To: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
Message-ID: <787610.94105.qm@web57616.mail.re1.yahoo.com>

Thanks Michael.

I will test drive it and report back. I will also be testing incoming fax and conference calls as well next week.

Cheers,

-Will




________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com> wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?
>
>I heard this was a feature that was implemented or going to be implemented in freeswitch.
>
>If it has been implemented, would you send me info on how to configure it?
>
>If it has not been implemented, would you state when it is planned to be implemented?
>
>I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report back.
-MC


      
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From lakindia89 at gmail.com  Fri Mar 19 01:27:07 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 19 Mar 2010 13:57:07 +0530
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
Message-ID: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>

Hi all,
I've updated to the latest SVN 17048.
Here is my dialplan:
<include>
        <context name="default">
                <extension name="outbound_soc">
                        <condition field="destination_number"
expression="^.*$">
                                <action application="ring_ready" />
                                <action application="set"
data="continue_on_fail=true"/>
                                <action application="set"
data="bypass_media=true"/>
                                <action application="set"
data="ignore_early_media=true"/>
                                <action application="set"
data="exec_after_bridge_app=park"/>
                                <action application="socket" data="
0.0.0.0:8447 async full"/>
                        </condition>
                </extension>
        </context>
</include>

>From a nc session I issued the following commands.
connect

sendmsg
call-command: execute
execute-app-name: answer

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: user/1010


I (user/1001) is able to hear the voice of the bridged party (user/1010).
But user/1010 is unable to hear my voice.

If I set the bypass_media=false, then both can hear the voice.

I wanted to know why when bypass_media=true, the user/1010 cannot hear my
voice.
Here is the freeswitch log:
http://pastebin.freeswitch.org/12484
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From jason at jasonjgw.net  Fri Mar 19 01:51:57 2010
From: jason at jasonjgw.net (Jason White)
Date: Fri, 19 Mar 2010 19:51:57 +1100
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
In-Reply-To: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
References: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
Message-ID: <20100319085157.GA18092@jdc.jasonjgw.net>

lakshmanan ganapathy <lakindia89 at gmail.com> wrote:
> I wanted to know why when bypass_media=true, the user/1010 cannot hear my
> voice.

It's probably due to NAT between the two end-points, or some other network
issue. Have you used Wireshark or a similar packet analysis tool to determine
whether RTP packets are being sent out by both end-points to the correct
addresses?

If they are, then you'll need to find out why the packets in one direction
fail to reach the destination.




From m.krivushin at imarto.net  Fri Mar 19 02:22:35 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Fri, 19 Mar 2010 15:22:35 +0600
Subject: [Freeswitch-users] Video pass problem
In-Reply-To: <FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>
References: <5be734a51002250459wb974018ue1ffd7d7a88ace59@mail.gmail.com>
	<FB2E4BDD-EF7A-43B3-B0CA-B7933CFDA623@jerris.com>
Message-ID: <5be734a51003190222w35f4db78he61ba6a138b41943@mail.gmail.com>

We use two different OSes - stable debian and bleeding edge ubuntu, but
problem live in both.
I think problem not in bleeding edge OS, but in way of processing rtp by FS.
I'm not sure, but think problem in renumerating of rtp packets in video
processing in mod_h26x.

If we have video problems with lost packets, after FS we have not this info,
its mascarade losts by renumerating. And, I suppose, decoding part cant
understand video with masacarding of lost frames.

All work fine with proxy mode, now, after small patching of old hack in
switch_rtp.c, tnx to anthm. But my billing module dont work in proxy mode
now, and I hear that proxy mode is hack for t.38.

2010/3/5 Michael Jerris <mike at jerris.com>

> double check if the behavior on this is the same with svn trunk of
> FreeSWITCH on something other than a bleeding edge operating system.
>
> Mike
>
> On Feb 25, 2010, at 7:59 AM, Mikhail Krivushin wrote:
>
> Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and
> see that we have 1280 video packets input, and only 560 passed to B leg.
> Anyone can point me to right direction? I can send pcap file by request.
>
> Most time I have black screen, and then one frame can appear, and I see
> frozen picture pair minutes, and then other frozen picture. We have not
> network issues, we have good video, when bypass FS. We have not perfomance
> troubles to.
>
> We have ubuntu 9.10 x64, and powerfull server board. uname:
> Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009
> x86_64 GNU/Linux
>
> config:
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h
> @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h
> @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From devel at thom.fr.eu.org  Fri Mar 19 03:51:13 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Fri, 19 Mar 2010 11:51:13 +0100
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
Message-ID: <e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>



Actually, I do not seem to have the same problem. I don't get the broken
pipe error, nor do I get any error in FS log or console (the error message
in log disappeared after some upgrade but I don't remember which).


However, what I can tell is that I get the problem as root AND as non
root. I can also tell that sending the exact same message with the exact
same command line as my freeswitch user work (sudo -u freeswitch cat
savedmail | sendmail -f freeswitch at mydomain -t some.user at mydomain). 

This
does not seem to be an environment problem either (I could not seem
significant differences between the 2). 

I'm about to strace what happens
too. 

Fran?ois 

On Thu, 18 Mar 2010 17:24:41 -0400, "Madovsky"  wrote: 
Ok, I read that there are some guys who had the same problem here. I run FS
as root, but sendmail as chroot smmsp user. You're right maybe it's a
thread child problem. I saw there is a windows condition already in your
code ? and for mac it should work since ssmtp is in standard in all
package, even for solaris and hp-ux. I will experiment with strace tonight
or tomorrow  ----- Original Message ----- FROM: Anthony Minessale [1] TO:
freeswitch-users at lists.freeswitch.org [2] SENT: Thursday, March 18, 2010
4:10 PM SUBJECT: Re: [Freeswitch-users] voicemail options [temporarly
solved]  The dilemma is we also need solution that is compat with windows,
bsd, mac etc.

That is sort of interesting that the pipe is breaking, it
would be nice to figure out why since not everybody
seems to have it.

I am
still wondering if it's a stack size running out issue in the child thread
from system.
Are you running FS as root or as a non-privileged user, i'm
wondering because maybe if its a regular user, its not able to increase the
stack size up to 8megs 

maybe you can push in strace into the command so
it will show why it breaks.

 On Thu, Mar 18, 2010 at 1:10 PM, Madovsky 
wrote:
   Also no need to uninstall sendmail, just install ssmtp and that's
it !   ----- Original Message ----- FROM: Madovsky [4] TO:
freeswitch-users at lists.freeswitch.org [5]    SENT: Thursday, March 18, 2010
1:59 PM SUBJECT: Re: [Freeswitch-users] voicemail options [temporarly
solved]  Hi Anthony,     after 5 days of tests to resolve the hell of
sendmail segmentation fault,   my conclusion is there is definitely a
problem with the combination of cat, pipe and sendmail or other mail app.  
I made a php script that fits the actual switch_utils.c exec, but it's the
same, it works one time on 10.   So the robustest solution is to replace
sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)      
configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the
switch.conf.xml like this        

and apply the patch attached (no choice
since it needs filename as argument and not STDIN).              hope it
will help people who have sendmail segmentation fault problem.      
Regards       Franck         ----- Original Message -----    FROM: Anthony
Minessale [6]    TO: freeswitch-users at lists.freeswitch.org [7]    SENT:
Wednesday, March 17, 2010 4:37 PM   SUBJECT: Re: [Freeswitch-users]
voicemail options [temporarly solved]   
 FYI if you made the change in
your working copy you could generate a patch with

svn diff >
patch.txt

After looking at your change, the problem is that many people
depend on this working the way it currently is.
Would you consider making a
script that mimics the current behavior of a mailer app and read the file
from stdin and deliver the subsequent file so it is backwards-compatible
with the way everything already works?

 On Wed, Mar 17, 2010 at 3:26 PM,
Madovsky  wrote:
     Sorry Anthony I never made an official update in C
language with diff patch   so the change is at the line 620, I commented
out the original line           ----- Original Message -----    FROM:
Anthony Minessale [9]    TO: freeswitch-users at lists.freeswitch.org [10]    
     SENT: Wednesday, March 17, 2010 3:55 PM   SUBJECT: Re:
[Freeswitch-users] voicemail options [temporarly solved]   
 could you get
a svn diff of the switch_utils.c so we can see what the change was?

 On
Wed, Mar 17, 2010 at 2:46 PM, Madovsky  wrote:
     Finally I succeed to
send voicemail with sendmail through a little php script (could be Perl
also)   it seems that sendmail 8.14.3 has a bug. But call it from PHP the
bug doesn't exist.       I attached my php script and switch_utils.c. Not
sure if the FS developers will accept my modification       Regards      
Franck     ----- Original Message -----    FROM: Madovsky [12]    TO:
freeswitch-users at lists.freeswitch.org [13]    SENT: Tuesday, March 16, 2010
4:35 PM   SUBJECT: Re: [Freeswitch-users] voicemail options   
   ok, I
tried 50 times to leave a message,   it worked only one time.   It seems
that sendmail needs to be free to send the email....         ----- Original
Message -----    FROM: devel at thom.fr.eu.org [14]    TO:
freeswitch-users at lists.freeswitch.org [15]    SENT: Tuesday, March 16, 2010
3:59 PM   SUBJECT: Re: [Freeswitch-users] voicemail options   

Are you
running debian, I have the same problem on debian x64 (already mentioned
here
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html
[16]) and cannot fix it.   

I don't know if it's a priviledge problem, as
I got the same results running FS with or without -u.   

Fran?ois   

DE :
freeswitch-users-bounces at lists.freeswitch.org [17]
[mailto:freeswitch-users-bounces at lists.freeswitch.org [18]] DE LA PART DE
Anthony Minessale
ENVOY : mardi 16 mars 2010 20:30
 :
freeswitch-users at lists.freeswitch.org [19]
OBJET : Re: [Freeswitch-users]
voicemail options    

are you running FS as root or a regular user?
maybe
there is a permission or resource limitation problem preventing it from
getting the larger stack space necessary?

On Tue, Mar 16, 2010 at 2:18 PM,
Madovsky  wrote:       

I noticed that sendmail makes a segfault when FS
is trying to send email      

and sendmail has some emails locked in
folder queue....      

Regards      

Franck    


_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[21]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[22]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[23]
http://www.freeswitch.org [24]    

-- 
Anthony Minessale
II

FreeSWITCH http://www.freeswitch.org/ [25]
ClueCon
http://www.cluecon.com/ [26]
Twitter: http://twitter.com/FreeSWITCH_wire
[27]

AIM: anthm
MSN:anthony_minessale at hotmail.com
[28]
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [29]
IRC:
irc.freenode.net [30] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[31]
iax:guest at conference.freeswitch.org/888
[32]
googletalk:conf+888 at conference.freeswitch.org [33]
pstn:+19193869900  


-------------------------


_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[34]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[35]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[36]
http://www.freeswitch.org
[37]

_______________________________________________
FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[38]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[39]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[40]
http://www.freeswitch.org [41]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [42]
ClueCon http://www.cluecon.com/
[43]
Twitter: http://twitter.com/FreeSWITCH_wire [44]

AIM:
anthm
MSN:anthony_minessale at hotmail.com
[45]
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [46]
IRC:
irc.freenode.net [47] #freeswitch

FreeSWITCH Developer
Conference
sip:888 at conference.freeswitch.org
[48]
iax:guest at conference.freeswitch.org/888
[49]
googletalk:conf+888 at conference.freeswitch.org
[50]
pstn:+19193869900

-------------------------


_______________________________________________
FreeSWITCH-users mailing
list
FreeSWITCH-users at lists.freeswitch.org
[51]
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[52]
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-- 
Anthony Minessale II

FreeSWITCH
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ClueCon http://www.cluecon.com/
[60]
Twitter: http://twitter.com/FreeSWITCH_wire [61]

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-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [76]
ClueCon http://www.cluecon.com/
[77]
Twitter: http://twitter.com/FreeSWITCH_wire [78]

AIM:
anthm
MSN:anthony_minessale at hotmail.com
[79]
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [80]
IRC:
irc.freenode.net [81] #freeswitch

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Conference
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[82]
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[83]
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Links:
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[1]
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[2]
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[3]
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[4] mailto:infos at madovsky.org
[5]
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[6]
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[7]
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[8]
mailto:infos at madovsky.org
[9] mailto:anthony.minessale at gmail.com
[10]
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[11]
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[12] mailto:infos at madovsky.org
[13]
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[14]
mailto:devel at thom.fr.eu.org
[15]
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[16]
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[17]
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[18]
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[19]
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[20]
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[21]
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[22]
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[23]
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[24]
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[25] http://www.freeswitch.org/
[26]
http://www.cluecon.com/
[27] http://twitter.com/FreeSWITCH_wire
[28]
mailto:MSN%3Aanthony_minessale at hotmail.com
[29]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[30]
http://irc.freenode.net
[31]
mailto:sip%3A888 at conference.freeswitch.org
[32]
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[33]
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[34]
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[35]
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[36]
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[37]
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[38]
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[39]
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[40]
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[41]
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[42] http://www.freeswitch.org/
[43]
http://www.cluecon.com/
[44] http://twitter.com/FreeSWITCH_wire
[45]
mailto:MSN%3Aanthony_minessale at hotmail.com
[46]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[47]
http://irc.freenode.net
[48]
mailto:sip%3A888 at conference.freeswitch.org
[49]
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[50]
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[51]
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[52]
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[53]
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[54]
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[55]
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[56]
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[57]
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[58]
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[59] http://www.freeswitch.org/
[60]
http://www.cluecon.com/
[61] http://twitter.com/FreeSWITCH_wire
[62]
mailto:MSN%3Aanthony_minessale at hotmail.com
[63]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[64]
http://irc.freenode.net
[65]
mailto:sip%3A888 at conference.freeswitch.org
[66]
http://iax:guest at conference.freeswitch.org/888
[67]
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[68]
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[69]
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[70]
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[71]
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[72]
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[73]
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[74]
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[75]
http://www.freeswitch.org
[76] http://www.freeswitch.org/
[77]
http://www.cluecon.com/
[78] http://twitter.com/FreeSWITCH_wire
[79]
mailto:MSN%3Aanthony_minessale at hotmail.com
[80]
mailto:PAYPAL%3Aanthony.minessale at gmail.com
[81]
http://irc.freenode.net
[82]
mailto:sip%3A888 at conference.freeswitch.org
[83]
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[84]
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From infos at madovsky.org  Fri Mar 19 04:41:30 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 07:41:30 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
Message-ID: <49B3842E181347948484F22AB2FABBD0@MOBILEE1705>

I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)
  ----- Original Message ----- 
  From: Fran?ois Legal 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 6:51 AM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  Actually, I do not seem to have the same problem. I don't get the broken pipe error, nor do I get any error in FS log or console (the error message in log disappeared after some upgrade but I don't remember which).

  However, what I can tell is that I get the problem as root AND as non root. I can also tell that sending the exact same message with the exact same command line as my freeswitch user work (sudo -u freeswitch cat savedmail | sendmail -f freeswitch at mydomain -t some.user at mydomain).

  This does not seem to be an environment problem either (I could not seem significant differences between the 2).

  I'm about to strace what happens too.



  Fran?ois



  On Thu, 18 Mar 2010 17:24:41 -0400, "Madovsky" <infos at madovsky.org> wrote:

    Ok, I read that there are some guys who had the same problem here.
    I run FS as root, but sendmail as chroot smmsp user. You're right maybe
    it's a thread child problem. I saw there is a windows condition already in your code ?
    and for mac it should work since ssmtp is in standard in all package, even for solaris and hp-ux.
    I will experiment with strace tonight or tomorrow
      ----- Original Message -----
      From: Anthony Minessale
      To: freeswitch-users at lists.freeswitch.org
      Sent: Thursday, March 18, 2010 4:10 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]
      The dilemma is we also need solution that is compat with windows, bsd, mac etc.

      That is sort of interesting that the pipe is breaking, it would be nice to figure out why since not everybody
      seems to have it.

      I am still wondering if it's a stack size running out issue in the child thread from system.
      Are you running FS as root or as a non-privileged user, i'm wondering because maybe if its a regular user, its not able to increase the stack size up to 8megs 

      maybe you can push in strace into the command so it will show why it breaks.




      On Thu, Mar 18, 2010 at 1:10 PM, Madovsky <infos at madovsky.org> wrote:

        Also no need to uninstall sendmail, just install ssmtp and that's it !
          ----- Original Message -----
          From: Madovsky
          To: freeswitch-users at lists.freeswitch.org
          Sent: Thursday, March 18, 2010 1:59 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]
          Hi Anthony,

          after 5 days of tests to resolve the hell of sendmail segmentation fault,
          my conclusion is there is definitely a problem with the combination of cat, pipe and sendmail or other mail app.
          I made a php script that fits the actual switch_utils.c exec, but it's the same, it works one time on 10.
          So the robustest solution  is to replace sendmail by ssmtp (yum install ssmtp or apt-get ssmtp etc...)

          configure /etc/ssmtp/revaliases and /etc/ssmtp/ssmtp.conf, change the switch.conf.xml like this

              
              

          and apply the patch attached (no choice since it needs filename as argument and not STDIN).
           
        hope it will help people who have sendmail segmentation fault problem.

        Regards

        Franck

          ----- Original Message ----- 
          From: Anthony Minessale 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Wednesday, March 17, 2010 4:37 PM
          Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


          FYI if you made the change in your working copy you could generate a patch with

          svn diff > patch.txt

          After looking at your change, the problem is that many people depend on this working the way it currently is.
          Would you consider making a script that mimics the current behavior of a mailer app and read the file from stdin and deliver the subsequent file so it is backwards-compatible with the way everything already works?



          On Wed, Mar 17, 2010 at 3:26 PM, Madovsky <infos at madovsky.org> wrote:

            Sorry Anthony I never made an official update in C language with diff patch
            so the change is at the line 620, I commented out the original line

              ----- Original Message ----- 
              From: Anthony Minessale 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Wednesday, March 17, 2010 3:55 PM
              Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


              could you get a svn diff of the switch_utils.c so we can  see what the change was?




              On Wed, Mar 17, 2010 at 2:46 PM, Madovsky <infos at madovsky.org> wrote:

                Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also)
                it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist.

                I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification

                Regards

                Franck
                  ----- Original Message ----- 
                  From: Madovsky 
                  To: freeswitch-users at lists.freeswitch.org 
                  Sent: Tuesday, March 16, 2010 4:35 PM
                  Subject: Re: [Freeswitch-users] voicemail options


                  ok, I tried 50 times to leave a message,
                  it worked only one time.
                  It seems that sendmail needs to be free to send the email....

                    ----- Original Message ----- 
                    From: devel at thom.fr.eu.org 
                    To: freeswitch-users at lists.freeswitch.org 
                    Sent: Tuesday, March 16, 2010 3:59 PM
                    Subject: Re: [Freeswitch-users] voicemail options


                    Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.html) and cannot fix it.

                    I don?t know if it?s a priviledge problem, as I got the same results running FS with or without ?u.



                    Fran?ois





                    De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Anthony Minessale
                    Envoy? : mardi 16 mars 2010 20:30
                    ? : freeswitch-users at lists.freeswitch.org
                    Objet : Re: [Freeswitch-users] voicemail options




                    are you running FS as root or a regular user?
                    maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary?



                    On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos at madovsky.org> wrote:

                    I noticed that sendmail makes a segfault when FS is trying to send email

                    and sendmail has some emails locked in folder queue....




                    Regards




                    Franck


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                    -- 
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                    FreeSWITCH http://www.freeswitch.org/
                    ClueCon http://www.cluecon.com/
                    Twitter: http://twitter.com/FreeSWITCH_wire

                    AIM: anthm
                    MSN:anthony_minessale at hotmail.com
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              -- 
              Anthony Minessale II

              FreeSWITCH http://www.freeswitch.org/
              ClueCon http://www.cluecon.com/
              Twitter: http://twitter.com/FreeSWITCH_wire

              AIM: anthm
              MSN:anthony_minessale at hotmail.com
              GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
              IRC: irc.freenode.net #freeswitch

              FreeSWITCH Developer Conference
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          -- 
          Anthony Minessale II

          FreeSWITCH http://www.freeswitch.org/
          ClueCon http://www.cluecon.com/
          Twitter: http://twitter.com/FreeSWITCH_wire

          AIM: anthm
          MSN:anthony_minessale at hotmail.com
          GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
          IRC: irc.freenode.net #freeswitch

          FreeSWITCH Developer Conference
          sip:888 at conference.freeswitch.org
          iax:guest at conference.freeswitch.org/888
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      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900





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From rentmycoder at gmail.com  Fri Mar 19 06:01:11 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Fri, 19 Mar 2010 14:01:11 +0100
Subject: [Freeswitch-users] how to export variables from b-leg to a-leg?
Message-ID: <50e456911003190601i395d108cg55c98a214e84eeae@mail.gmail.com>

Hi guys,
Freeswitch Rocks!

Does anybody know how to export variables from the b-leg to a-leg...
I start a bridge from lua on an outgoing gateway and on the answer of
the b-leg I would like to set a variable on the a-leg...

<action application="export" data="nolocal:execute_on_answer=lua
cc.lua extensions_answer ${uuid}"/>
<bridge ....>

from cc.lua I try:
session:execute("export", "foo=123");

but no success...
any hint?



From brian at freeswitch.org  Fri Mar 19 06:20:59 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 08:20:59 -0500
Subject: [Freeswitch-users] ProxyMedia Unable to hear voice
In-Reply-To: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
References: <7d79b3931003190127o693cec26u1ee55f2653f7d0da@mail.gmail.com>
Message-ID: <D5BECA6B-086E-45AC-940A-D48DB39C35B2@freeswitch.org>

Because you can't use Proxy Media in this case.  You do realize that FreeSWITCH by default will proxy the media thru the switch.  Proxy media is a different ball of wax and has exact ONE USE CASE.  Anyone using Proxy media and isn't doing T.38/UDPTL needs to stop using it.  Its only for T.38 passthru so that FreeSWITCH doesn't look at the contents of the packet and just passes them along.

1. Default Media Passes thru FreeSWITCH and parses the RTP packets up into the switch.

2. Proxy Media passes thru FreeSWITCH but FreeSWITCH can't touch the media because it could be formats we don't know about ie UDPTL

3. Bypass Media keeps the signaling on FreeSWITCH but makes the media go P2P.


The bottom line is YOU SHOULDN'T BE USING PROXY MEDIA.

/b

On Mar 19, 2010, at 3:27 AM, lakshmanan ganapathy wrote:

> Hi all,
> I've updated to the latest SVN 17048.
> Here is my dialplan:
> <include>
>         <context name="default">
>                 <extension name="outbound_soc">
>                         <condition field="destination_number" expression="^.*$">
>                                 <action application="ring_ready" />
>                                 <action application="set" data="continue_on_fail=true"/>
>                                 <action application="set" data="bypass_media=true"/>
>                                 <action application="set" data="ignore_early_media=true"/>
>                                 <action application="set" data="exec_after_bridge_app=park"/>
>                                 <action application="socket" data="0.0.0.0:8447 async full"/>
>                         </condition>
>                 </extension>
>         </context>
> </include>
> 
> From a nc session I issued the following commands.
> connect
> 
> sendmsg
> call-command: execute
> execute-app-name: answer
> 
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: user/1010
> 
> 
> I (user/1001) is able to hear the voice of the bridged party (user/1010). But user/1010 is unable to hear my voice.
> 
> If I set the bypass_media=false, then both can hear the voice.
> 
> I wanted to know why when bypass_media=true, the user/1010 cannot hear my voice.
> Here is the freeswitch log:
> http://pastebin.freeswitch.org/12484
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From testeador01 at gmail.com  Fri Mar 19 06:39:37 2010
From: testeador01 at gmail.com (Milena)
Date: Fri, 19 Mar 2010 08:39:37 -0500
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
In-Reply-To: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
References: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
Message-ID: <c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>

Hello,

None of them are ok for the domain you want to register to, when you set up
multiple domains, you have to return the results with the domain name you're
giving information about, make it return another xml with
<domain name="192.168.101.102">

2010/3/18 lloyd thomas <lloydie.t at googlemail.com>

> I eventually managed to get mod_xml_curl working for me. Now I trying to be
> a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
> wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I
> get an error when a phone tries to register.
>
> 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
> 203 at 192.168.101.102]
> You must define a domain called '192.168.101.102' in your directory and add
> a user with the id="203" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
> 192.168.91.28
>
> I think I am nearly there,but I suspect there is something wrong with the
> resulting 'user directory' xml.
> ---------------------------------------------------------------
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="User Directory">
> <domain name="default">
> <groups>
>   <group name="default">
>   <users>
>   <user id="203">
>     <params>
>       <param name="password" value="$${default_password}"/>
>       <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>       <param name="vm-password" value="1234"/>
>       <param name="vm-disk-quota" value="30"/>
>     </params>
>     <variables>
>       <variable name="user_context" value="default"/>
>       <variable name="accountcode" value="203"/>
>       <variable name="day_bar" value="3"/>
>       <variable name="night_bar" value="3"/>
>       <variable name="effective_caller_id_name" value="Extension 203"/>
>       <variable name="effective_caller_id_number" value="203"/>
>       <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number"
> value="$${outbound_caller_number}"/>
>     </variables>
>   </user>
>   </users>
>  </group>
> </groups>
> </domain>
> </section>
> </document>
> --------------------------------------------------------------------
>
> I think the Company Directory XML is OK
> --------------------------------------------------------------------
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
> <section name="directory" description="Company Directory">
> <domain name="default">
>   <params>
>     <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>   </params>
>   <variables>
>     <variable name="record_stereo" value="true"/>
>     <variable name="default_gateway" value="$${default_provider}"/>
>     <variable name="default_areacode" value="$${default_areacode}"/>
>     <variable name="transfer_fallback_extension" value="operator"/>
>     <variable name="user_context" value="default"/>
>   </variables>
> </domain>
> <domain name="phisystems.dom">
>   <params>
>     <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>   </params>
>   <variables>
>     <variable name="record_stereo" value="true"/>
>     <variable name="default_gateway" value="$${default_provider}"/>
>     <variable name="default_areacode" value="$${default_areacode}"/>
>     <variable name="transfer_fallback_extension" value="operator"/>
>     <variable name="user_context" value="phisystems.dom"/>
>   </variables>
> </domain>
> </section>
> </document>
> --------------------------------------------------------------
>
> I get the following result when I 'sofia status'
>                      Name          Type
> Data     State
>
> =================================================================================================
>                  internal       profile
> sip:mod_sofia at 192.168.101.102:5060   RUNNING (0)
>            phisystems.dom         alias
> internal     ALIASED
>             internal-ipv6       profile           sip:mod_sofia@[::1]:5060
> RUNNING (0)
>                   default         alias
> internal     ALIASED
>                  external       profile
> sip:mod_sofia at 192.168.101.102:5080   RUNNING (0)
>
> =================================================================================================
>
> Thanks in advance
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Fri Mar 19 07:24:40 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 19 Mar 2010 10:24:40 -0400
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <AD5B87549E604DD3BBD73065A1EF6317@cune.pri>
References: <82dae1c01003181328o384adbax887b94663523288c@mail.gmail.com>
	<AD5B87549E604DD3BBD73065A1EF6317@cune.pri>
Message-ID: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>

On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann <Russell.Mosemann at cune.org
> wrote:

>
> Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all
> that bad here, but it would be nice to use EC.
>
>
You can always get a Sangoma board with HWEC :-)

-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From moises.silva at gmail.com  Fri Mar 19 07:30:43 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 19 Mar 2010 10:30:43 -0400
Subject: [Freeswitch-users] openzap no audio?
In-Reply-To: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
References: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
Message-ID: <c4d05cbe1003190730k9bf6802vcc25a362c64d2edd@mail.gmail.com>

Hello,

People have reported problems with audio using DAHDI with software EC. If
you have it enabled, disable it and see if that helps ( you may have echo
though ).

Few ( if any ) of the FreeSWITCH developers use Digium hardware, most use
Sangoma, if you want something that works and is completely supported go
with Sangoma then.

On Thu, Mar 18, 2010 at 1:07 AM, Tom Christensen <pavera at gmail.com> wrote:

> I've got freeswitch installed with a digium te110p hooked up to a PRI.
>
> calls through the system work fine SIP->SIP, calling and receiving calls
> over the PRI works, (IE the signalling part, a call can be made and incoming
> calls come in over the PRI and I can route them based on DID), however,
> there is no audio to either party.  I've checked, and there is an RTP stream
> running while the call is up, there is no reason it would be blocked (IE,
> SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
> few posts that talked about no audio for a few seconds after a call is
> connected, or that calls would connect and then drop after a few seconds.
>  This is different, I can leave the call up for many minutes, it doesn't
> drop, and audio never starts coming through.
>
> I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
> the latest release of zaptel first, freeswitch wouldn't even load with that.
>
> I'm going to try libpri now I think as I've exhausted all the other options
> I can think of.
>
> Any input/ideas would be greatly appreciated.
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Russell.Mosemann at cune.org  Fri Mar 19 07:49:38 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 19 Mar 2010 14:49:38 -0000
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>
Message-ID: <20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>

Moises Silva <moises.silva at gmail.com> said:
> 
> On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann
<Russell.Mosemann at cune.org
> > wrote:
> 
> >
> > Sorry, I'm looking for a solution, myself. The echo over the T1 isn't all
> > that bad here, but it would be nice to use EC.
> >
> >
> You can always get a Sangoma board with HWEC :-)

I'd love to, believe me. In the past, we had four Sangoma data T1 boards
and load balanced our connection to the Internet over them. The hardware
was solid, setup was pretty easy and I got quick answers from support. It
was great.

Unfortunately, there is absolutely no money for the current project. We
had to scrounge a throwaway server and buy a cheap PRI T1 card off of
Ebay. We happened to acquire an additional T1 license on our Siemens
telephone switch as part of another project, and we bought a used T1 card
for the telephone switch off of Ebay. We've only spent about $100 on
something that could easily cost $10,000. We're hoping that after we get
a proof of concept going, the right people will see the value and fund a
better solution. We'll keep you mind. ;-)

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From lloydie.t at googlemail.com  Fri Mar 19 07:53:10 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Fri, 19 Mar 2010 14:53:10 +0000
Subject: [Freeswitch-users] Newbie problem with Multiple Tenants and
	mod_xml_curl
In-Reply-To: <c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>
References: <8b61bd671003181445o53b5592bl1ff74aea12e7d65a@mail.gmail.com>
	<c29b930b1003190639v3b758009nfcf273f814d9c974@mail.gmail.com>
Message-ID: <8b61bd671003190753t506d0c63t6b92c032e3830400@mail.gmail.com>

Thanks Milena. I think I have a much better understanding of how it works,
the main thing being you have to have an entry in the DNS server to point
the multi-tenancy domain name to the IP address of the FS server and make
sure the SIP client also has the correct domain name. All is now better with
the following config (unless you can see a glaring error).
User Directory
------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="User Directory">
<domain name="$${domain}">
<groups>
  <group name="default">
  <users>
  <user id="203">
    <params>
      <param name="password" value="$${default_password}"/>
      <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      <param name="vm-password" value="1234"/>
      <param name="vm-disk-quota" value="30"/>
    </params>
    <variables>
      <variable name="user_context" value="default"/>
      <variable name="accountcode" value="203"/>
      <variable name="day_bar" value="3"/>
      <variable name="night_bar" value="3"/>
      <variable name="effective_caller_id_name" value="Extension 203"/>
      <variable name="effective_caller_id_number" value="203"/>
      <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number"
value="$${outbound_caller_number}"/>
    </variables>
  </user>
  </users>
 </group>
</groups>
</domain>
</section>
</document>
--------------------------------------------------

Company Directory
------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory" description="Company Directory">
<domain name="$${domain}">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="default"/>
  </variables>
</domain>
<domain name="phitel.phisys.dom">
  <params>
    <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
  </params>
  <variables>
    <variable name="record_stereo" value="true"/>
    <variable name="default_gateway" value="$${default_provider}"/>
    <variable name="default_areacode" value="$${default_areacode}"/>
    <variable name="transfer_fallback_extension" value="operator"/>
    <variable name="user_context" value="phitel.phisys.dom"/>
  </variables>
</domain>
</section>
</document>
--------------------------------------------------------

On 19 March 2010 13:39, Milena <testeador01 at gmail.com> wrote:

> Hello,
>
> None of them are ok for the domain you want to register to, when you set up
> multiple domains, you have to return the results with the domain name you're
> giving information about, make it return another xml with
> <domain name="192.168.101.102">
>
> 2010/3/18 lloyd thomas <lloydie.t at googlemail.com>
>
>> I eventually managed to get mod_xml_curl working for me. Now I trying to
>> be a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the
>> wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But
>> I get an error when a phone tries to register.
>>
>> 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [
>> 203 at 192.168.101.102]
>> You must define a domain called '192.168.101.102' in your directory and
>> add a user with the id="203" attribute
>> and you must configure your device to use the proper domain in it's
>> authentication credentials.
>> 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure
>> (REGISTER) on sofia profile 'internal' for [203 at 192.168.101.102] from ip
>> 192.168.91.28
>>
>> I think I am nearly there,but I suspect there is something wrong with the
>> resulting 'user directory' xml.
>> ---------------------------------------------------------------
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="directory" description="User Directory">
>> <domain name="default">
>> <groups>
>>   <group name="default">
>>   <users>
>>   <user id="203">
>>     <params>
>>       <param name="password" value="$${default_password}"/>
>>       <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>       <param name="vm-password" value="1234"/>
>>       <param name="vm-disk-quota" value="30"/>
>>     </params>
>>     <variables>
>>       <variable name="user_context" value="default"/>
>>       <variable name="accountcode" value="203"/>
>>       <variable name="day_bar" value="3"/>
>>       <variable name="night_bar" value="3"/>
>>       <variable name="effective_caller_id_name" value="Extension 203"/>
>>       <variable name="effective_caller_id_number" value="203"/>
>>       <variable name="outbound_caller_id_name"
>> value="$${outbound_caller_name}"/>
>>       <variable name="outbound_caller_id_number"
>> value="$${outbound_caller_number}"/>
>>     </variables>
>>   </user>
>>   </users>
>>  </group>
>> </groups>
>> </domain>
>> </section>
>> </document>
>> --------------------------------------------------------------------
>>
>> I think the Company Directory XML is OK
>> --------------------------------------------------------------------
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>> <section name="directory" description="Company Directory">
>> <domain name="default">
>>   <params>
>>     <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>   </params>
>>   <variables>
>>     <variable name="record_stereo" value="true"/>
>>     <variable name="default_gateway" value="$${default_provider}"/>
>>     <variable name="default_areacode" value="$${default_areacode}"/>
>>     <variable name="transfer_fallback_extension" value="operator"/>
>>     <variable name="user_context" value="default"/>
>>   </variables>
>> </domain>
>> <domain name="phisystems.dom">
>>   <params>
>>     <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>   </params>
>>   <variables>
>>     <variable name="record_stereo" value="true"/>
>>     <variable name="default_gateway" value="$${default_provider}"/>
>>     <variable name="default_areacode" value="$${default_areacode}"/>
>>     <variable name="transfer_fallback_extension" value="operator"/>
>>     <variable name="user_context" value="phisystems.dom"/>
>>   </variables>
>> </domain>
>> </section>
>> </document>
>> --------------------------------------------------------------
>>
>> I get the following result when I 'sofia status'
>>                      Name          Type
>> Data     State
>>
>> =================================================================================================
>>                  internal       profile
>> sip:mod_sofia at 192.168.101.102:5060   RUNNING (0)
>>            phisystems.dom         alias
>> internal     ALIASED
>>             internal-ipv6       profile           sip:mod_sofia@[::1]:5060
>> RUNNING (0)
>>                   default         alias
>> internal     ALIASED
>>                  external       profile
>> sip:mod_sofia at 192.168.101.102:5080   RUNNING (0)
>>
>> =================================================================================================
>>
>> Thanks in advance
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Fri Mar 19 08:03:45 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Fri, 19 Mar 2010 11:03:45 -0400
Subject: [Freeswitch-users] make errors - mod_cepstral
Message-ID: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>

Hi All,

I am trying to enable flite & cepstral on a existing
FreeSWITCH installation. I follow the following steps

1. build directory  cp modules.con.in modules.conf

2. edit modules.conf
remove the comments from the following

asr_tts/mod_flite
asr_tts/mod_cepstral

3../bootstrap.sh

4. ./configure

5. make

I get the following error

make[5]: *** [mod_cepstral.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1

How can I successfully enable the mod_cepstral and flite.



Thanks
Lloyd
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From brian at freeswitch.org  Fri Mar 19 08:09:46 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 10:09:46 -0500
Subject: [Freeswitch-users] make errors - mod_cepstral
In-Reply-To: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>
References: <8a19bf2e1003190803pe5b5c38h2dff6d8c71130f89@mail.gmail.com>
Message-ID: <A767445F-85A7-42D4-82DC-93561665502E@freeswitch.org>

chances are you don't have cepstral installed which is required to use/compile it.

/b

On Mar 19, 2010, at 10:03 AM, Aloysius Lloyd wrote:

> make[5]: *** [mod_cepstral.lo] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_cepstral-all] Error 1
> make[2]: *** [all-recursive] Error 1




From fs-list at communicatefreely.net  Fri Mar 19 08:12:27 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 11:12:27 -0400
Subject: [Freeswitch-users] FS Best practices
In-Reply-To: <c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>
References: <BLU110-DS11B84F16B12464595A3B07A72B0@phx.gbl>
	<7b197bef1003180533n4182b0b0re5f255813140093e@mail.gmail.com>
	<87f2f3b91003180829k34105b60k9169878afc940b2c@mail.gmail.com>
	<021d01cac6bb$5d73e1e0$185ba5a0$@net>	<BLU110-DS5740D8147B1F68C12979BA72B0@phx.gbl>
	<8CA9D27B-75CA-4856-B05E-7E8F749D3A79@freeswitch.org>
	<c32ee1cf1003181345i6b695200j9d1fa9b10a0cb037@mail.gmail.com>
Message-ID: <4BA3945B.30104@communicatefreely.net>

I think there are advantages, at least from a functionality standpoint.

If you have a cluster setup, the core DB can be shared across FS machines.  This gives each machine
access to things like presence, voice mail counts and settings, and the mod_limit counts can be
global, as opposed to per-machine numbers.  This is handy if you want to enforce a channel count for
 a customer.  You can also share SIP registrations across the cluster.  I haven't tried this too
much yet, but it should be possible for a phone to register to machine A.  If machine B needs to
send a call to the phone, it has the contact data and can send the call to the phone directly (as
long as there are no NAT issues here).  I don't know how often each machine polls the DB for
presence information.  There might be a lag for things like BLF lamps across the cluster, although I
think there is another way to solve that.

You can also access the data in the DB with your PHP scripts for status monitoring and other useful
things.  I have heard that altering this data is a bad idea, but running queries on it can be handy.
   You can make a nice web dashboard that will give you all registered endpoints, current SIP
dialogs, channel counts (via mod_limit), voice mail counts, etc.  By accessing the voice mail
tables, you can also make an XML "visual voicemail" interface for phones like the Aastra 55i and 57i.

Hope that's helpful.

-Tim

Fraser Redmond wrote:
> Is there any advantage at all to putting the core database on your own
> external db? Or is it just a control freak thing? I was looking at this
> the other day, and only ever read that it was possible, not anything
> about why you'd want to do it.
> 
> Cheers,
> Fraser
> 
> 
> 
> 
> On Thu, Mar 18, 2010 at 5:41 PM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>      Its really up to you it works both ways.  But I would recommend a
>     dedicated network for interconnection and management and not run
>     that over the public/sip facing interfaces.
> 
>     /b
> 
>     On Mar 18, 2010, at 12:37 PM, Todd wrote:
> 
>>     Thanks folks!  I believe we will be doing most of our coding in
>>     php.  We are also going to have independent DB servers?. I was
>>     also wondering if putting the FS database on a DB server is a god
>>     idea, or if it should be left on the FS box.  ?Preciate it yall. 
>>     Learning is phun
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From Suneel.Papineni at mettoni.com  Fri Mar 19 08:20:06 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 15:20:06 -0000
Subject: [Freeswitch-users] Conference with FSComm
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>

Hi,

 

I am trying to achieve 3-way (or 3-party) conference scenario using
FSComm. Scenario is as follows:

 

An incoming call comes to FSComm from an extension 'A' and is answered.
Now FSComm makes a call to another extension 'B' and is answered. Then I
want to bridge all three calls and be in conference. So all three
parties A, B and FSComm will be in conference.

 

Could someone please give me an idea how can I achieve this scenario
using Event Sockets. (want to test with fs_cli using commands).

 

Thanks & Regards

Suneel

 


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From kenfulmer at icstechnologysolutions.com  Fri Mar 19 08:13:34 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Fri, 19 Mar 2010 10:13:34 -0500
Subject: [Freeswitch-users] Newbie Questions
Message-ID: <00a701cac776$bf147de0$3d3d79a0$@com>

1.       We would like to transcode from g711mu internally to g729 for PSTN
calls. Is this possible? 

2.       We are using sipX as an open source PBX. We'd like to use
FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
phones use SIP REFER messages to do call transfers and our ITSPs can't
handle those. So, we've tried the "under the hood" FreeSwitch within sipX.
It works for many applications but we need to scale a unified system for
larger customer deployments. If we want to create a trunk to an ITSP and a
trunk to an internal PBX, would we use the external gateway XML files for
both? We don't plan to point any internal phones to the FreeSwitch box
(we're using sipX as an internal PBX). 

3.       If we use multiple ITSPs, can we assign a priority to each one, so
one will be chosen first? 

4.       I understand no performance metrics are posted. Can anyone share
their experiences with passing RTP through the server? Does RTP affect
performance in a big way (very open ended question, I know). We are looking
for ball park answers so we'll have some idea of scalability for our
designs. 

 

Thanks,

 

Ken Fulmer

 

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From robert.hadley at teotech.com  Fri Mar 19 09:10:08 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 09:10:08 -0700
Subject: [Freeswitch-users] How does mod_directory get users information?
Message-ID: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>

Hello,

 

Where does mod_directory get the user information it uses to put into the
database in creates?

 

Thanks,

Robert

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From infos at madovsky.org  Fri Mar 19 09:29:10 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 12:29:10 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <F1775BE5441B4234B9A127CD3F6AC2A8@MOBILEE1705>

I set FS internal on port 5080 and external 5060.
2 softphones on public IP, so no nat.
the sip invite is ok, but there's a media timeout..
port 5080 and 5060 are opened on my firewall
must I set any nat params anyway ?

Thanks

Franck
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From fs-list at communicatefreely.net  Fri Mar 19 09:57:14 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 12:57:14 -0400
Subject: [Freeswitch-users] How does mod_directory get users information?
In-Reply-To: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
References: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
Message-ID: <4BA3ACEA.1040503@communicatefreely.net>

It looks in the user directory.

It will request users in the domain specified when you invoke it.

Something I learned (with help from the author), was that the users have to be in a group.  The
default group will do.

If you are using xml_curl to generate your directory dynamically, you have to set your script up so
that it will return all users in the domain when the user isn't specified in the query.  You don't
have to return all the variables though, just the user's name and number will usually work.

Robert Hadley wrote:
> Hello,
> 
>  
> 
> Where does mod_directory get the user information it uses to put into
> the database in creates?
> 
>  
> 
> Thanks,
> 
> Robert
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From infos at madovsky.org  Fri Mar 19 10:05:04 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:05:04 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>

with bypass-media=true everything is ok, 
but I need to pass media through FS, I missed
some settings but don't know where...
I use default configs, inverted the internal and external port
and commented out all nat params.

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 12:29 PM
  Subject: rtp timeout issue


  I set FS internal on port 5080 and external 5060.
  2 softphones on public IP, so no nat.
  the sip invite is ok, but there's a media timeout..
  port 5080 and 5060 are opened on my firewall
  must I set any nat params anyway ?

  Thanks

  Franck
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From anthony.minessale at gmail.com  Fri Mar 19 10:17:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:17:17 -0500
Subject: [Freeswitch-users] rtp timeout issue
In-Reply-To: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
Message-ID: <191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>

you should also unblock the rtp port range
16384-32768 udp


On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

>  with bypass-media=true everything is ok,
> but I need to pass media through FS, I missed
> some settings but don't know where...
> I use default configs, inverted the internal and external port
> and commented out all nat params.
>
> Thanks
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 12:29 PM
> *Subject:* rtp timeout issue
>
> I set FS internal on port 5080 and external 5060.
> 2 softphones on public IP, so no nat.
> the sip invite is ok, but there's a media timeout..
> port 5080 and 5060 are opened on my firewall
> must I set any nat params anyway ?
>
> Thanks
>
> Franck
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 19 10:21:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:21:46 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <00a701cac776$bf147de0$3d3d79a0$@com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
Message-ID: <191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>

If you plan to transcode g729 to g711 it will impact performance
dramatically but you should still get a few hundered channels up on a nice
server box.  You could contact consulting at freeswitch.org about g729 license
options.

With no g729 you could scale a nice 8 core box into a few thousand channels
pushing rtp.

Do you have a requirement to transcode g729?  you could just use g711 right
from the phone couldn't you?


On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  1.       We would like to transcode from g711mu internally to g729 for
> PSTN calls. Is this possible?
>
> 2.       We are using sipX as an open source PBX. We?d like to use
> FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
> phones use SIP REFER messages to do call transfers and our ITSPs can?t
> handle those. So, we?ve tried the ?under the hood? FreeSwitch within sipX.
> It works for many applications but we need to scale a unified system for
> larger customer deployments. If we want to create a trunk to an ITSP and a
> trunk to an internal PBX, would we use the external gateway XML files for
> both? We don?t plan to point any internal phones to the FreeSwitch box
> (we?re using sipX as an internal PBX).
>
> 3.       If we use multiple ITSPs, can we assign a priority to each one,
> so one will be chosen first?
>
> 4.       I understand no performance metrics are posted. Can anyone share
> their experiences with passing RTP through the server? Does RTP affect
> performance in a big way (very open ended question, I know). We are looking
> for ball park answers so we?ll have some idea of scalability for our
> designs.
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Fri Mar 19 10:29:35 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:29:35 -0400
Subject: [Freeswitch-users] rtp timeout issue
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
	<191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>
Message-ID: <56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>

yes I did it also.
the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:17 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  you should also unblock the rtp port range
  16384-32768 udp



  On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

    with bypass-media=true everything is ok, 
    but I need to pass media through FS, I missed
    some settings but don't know where...
    I use default configs, inverted the internal and external port
    and commented out all nat params.

    Thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, March 19, 2010 12:29 PM
      Subject: rtp timeout issue


      I set FS internal on port 5080 and external 5060.
      2 softphones on public IP, so no nat.
      the sip invite is ok, but there's a media timeout..
      port 5080 and 5060 are opened on my firewall
      must I set any nat params anyway ?

      Thanks

      Franck

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Fri Mar 19 10:42:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:42:33 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>
References: <c4d05cbe1003190724l6e9032c8l7623d4f2b24a0130@mail.gmail.com>
	<20100319144938.D9E2D2C5AF4@cuneorg-email.cune.pri>
Message-ID: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>

Look at it this way.

The PRI signaling is a data channel, it's got nothing to do with echo
cancelers.  The echo can is for the audio legs.
The OSLEC is a kernel module and dahdi is Digiums driver also a kernel
module. OpenZAP only opens these devices and uses them as advertised.

Keep in mind, When we added support for digium cards it was when there was
still zaptel and every day they change things in their code which is mostly
kernel space and they have absolutely no interest in having it work on
FreeSWITCH, even though we are both open source applications they treat
FreeSWITCH like a threat.

You are trying to escape from asterisk but you are still using mostly their
technology.

So bottom line changing around which pri stack you use is not going to solve
audio problems.
All OpenZAP does with digium cards is open the device and read and write
from it.
If you are getting no audio the problem is happening in the kernel before
OpenZAP gets to see anything.


My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
there is a line
echo_cancel_level => 64

try erasing or commenting it out with a leading ;

>From my recollection when you use OSLEC, you have to disable the
dahdi/zaptel echo canceler.

This is the only setting I can think of that allows OpenZAP to influence the
dahdi kernel mod.

BTW, you got that card for $100 because it really is worth that much, maybe
less ;)

Everyone who uses Sangoma cards with OpenZAP have no audio problems at all.




On Fri, Mar 19, 2010 at 9:49 AM, <Russell.Mosemann at cune.org> wrote:

> Moises Silva <moises.silva at gmail.com> said:
> >
> > On Thu, Mar 18, 2010 at 8:48 PM, Russell Mosemann
> <Russell.Mosemann at cune.org
> > > wrote:
> >
> > >
> > > Sorry, I'm looking for a solution, myself. The echo over the T1 isn't
> all
> > > that bad here, but it would be nice to use EC.
> > >
> > >
> > You can always get a Sangoma board with HWEC :-)
>
> I'd love to, believe me. In the past, we had four Sangoma data T1 boards
> and load balanced our connection to the Internet over them. The hardware
> was solid, setup was pretty easy and I got quick answers from support. It
> was great.
>
> Unfortunately, there is absolutely no money for the current project. We
> had to scrounge a throwaway server and buy a cheap PRI T1 card off of
> Ebay. We happened to acquire an additional T1 license on our Siemens
> telephone switch as part of another project, and we bought a used T1 card
> for the telephone switch off of Ebay. We've only spent about $100 on
> something that could easily cost $10,000. We're hoping that after we get
> a proof of concept going, the right people will see the value and fund a
> better solution. We'll keep you mind. ;-)
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 19 10:44:09 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:44:09 -0500
Subject: [Freeswitch-users] rtp timeout issue
In-Reply-To: <56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705>
	<191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com>
	<56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
Message-ID: <191c3a031003191044n3334b4ebm5abe7ce52c9e799b@mail.gmail.com>

try stopping iptables for 1 call to make sure it is not related to that.



On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

>  yes I did it also.
> the issue is only noticeable if I use 2 softphones on the same public IP
> where there is FS.
> I tried from another PC (even through nat) and it works fine... Arrgh, I'm
> lost
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 1:17 PM
> *Subject:* Re: [Freeswitch-users] rtp timeout issue
>
> you should also unblock the rtp port range
> 16384-32768 udp
>
>
> On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  with bypass-media=true everything is ok,
>> but I need to pass media through FS, I missed
>> some settings but don't know where...
>> I use default configs, inverted the internal and external port
>> and commented out all nat params.
>>
>> Thanks
>>
>> ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, March 19, 2010 12:29 PM
>> *Subject:* rtp timeout issue
>>
>> I set FS internal on port 5080 and external 5060.
>> 2 softphones on public IP, so no nat.
>> the sip invite is ok, but there's a media timeout..
>> port 5080 and 5060 are opened on my firewall
>> must I set any nat params anyway ?
>>
>> Thanks
>>
>> Franck
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Fri Mar 19 10:53:18 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 13:53:18 -0400
Subject: [Freeswitch-users] rtp timeout issue
References: <4467B10D825D449A9B2BA670DF0837FB@MOBILEE1705><191c3a031003191017r3b08e462v4903447d860363e4@mail.gmail.com><56B5BDB382C4400D9AAE22CC8FF8DD52@MOBILEE1705>
	<191c3a031003191044n3334b4ebm5abe7ce52c9e799b@mail.gmail.com>
Message-ID: <87DAD62EECE64DF9B33E3089A3D680A1@MOBILEE1705>

bingo.... :), 
grrrr, now I need to read all my iptables :(
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:44 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  try stopping iptables for 1 call to make sure it is not related to that.




  On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

    yes I did it also.
    the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
    I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, March 19, 2010 1:17 PM
      Subject: Re: [Freeswitch-users] rtp timeout issue


      you should also unblock the rtp port range
      16384-32768 udp



      On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

        with bypass-media=true everything is ok, 
        but I need to pass media through FS, I missed
        some settings but don't know where...
        I use default configs, inverted the internal and external port
        and commented out all nat params.

        Thanks
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Friday, March 19, 2010 12:29 PM
          Subject: rtp timeout issue


          I set FS internal on port 5080 and external 5060.
          2 softphones on public IP, so no nat.
          the sip invite is ok, but there's a media timeout..
          port 5080 and 5060 are opened on my firewall
          must I set any nat params anyway ?

          Thanks

          Franck

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org





      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      iax:guest at conference.freeswitch.org/888
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Fri Mar 19 10:58:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 12:58:01 -0500
Subject: [Freeswitch-users] variables in gateway profile
In-Reply-To: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
Message-ID: <191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>

yes and you can specify which direction you want to set them on, "inbound",
"outbound" or "both"

<variables>
  <variable name="foo" data="bar" direction="inbound"/>
</variables>


On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:

>  Is it possible to create <variable> in gateway profiles like users
> profiles ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From nicolas at medularis.com  Fri Mar 19 11:19:34 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 19 Mar 2010 14:19:34 -0400
Subject: [Freeswitch-users] Originate timeout with multiple destinations
Message-ID: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>

Hi,

I need to do an originate to multiple destinations, but I need that the
first call has a timeout of 17 seconds, and if it's not answered, then call
the other destinations.

I was thinking something like:

originate
{ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx,
sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz


I tried using the call_timeout, timeout and originate_timeout variables in
the square brackets for the first dialstring, but it didn't work. Is there a
way of doing this with 1 originate command?


Thanks!
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From Suneel.Papineni at mettoni.com  Fri Mar 19 11:20:44 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Fri, 19 Mar 2010 18:20:44 -0000
Subject: [Freeswitch-users] [Freeswitch-dev] Conference with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
Message-ID: <3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>

I am not sure if the following method works. Please let me know how
could I implement this through event sockets.

 

'A' calls FSComm and FSComm calls 'B'. Now FSComm is having call details
of all call legs. If possible FSComm now creates a conference and join
all 3 call legs to the conference. This might solve the problem, but
could you please let me know how can create a conference and join all
the legs. Please let me know the commands for this. I checked with
"mod_conference" module, unable to find respective commands. I tried
"api conference....", it is saying command not found. I tried
"conference....." command, it is also not working.

 

Thanks & Regards

Suneel

 

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Suneel
Papineni
Sent: 19 March 2010 15:20
To: freeswitch-dev at lists.freeswitch.org
Cc: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-dev] Conference with FSComm

 

Hi,

 

I am trying to achieve 3-way (or 3-party) conference scenario using
FSComm. Scenario is as follows:

 

An incoming call comes to FSComm from an extension 'A' and is answered.
Now FSComm makes a call to another extension 'B' and is answered. Then I
want to bridge all three calls and be in conference. So all three
parties A, B and FSComm will be in conference.

 

Could someone please give me an idea how can I achieve this scenario
using Event Sockets. (want to test with fs_cli using commands).

 

Thanks & Regards

Suneel

 

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From Russell.Mosemann at cune.org  Fri Mar 19 11:21:29 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Fri, 19 Mar 2010 18:21:29 -0000
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
Message-ID: <20100319182129.49EE5129EA7@cuneorg-email.cune.pri>

Anthony Minessale <anthony.minessale at gmail.com> said:

> All OpenZAP does with digium cards is open the device and read
> and write from it. If you are getting no audio the problem is
> happening in the kernel before OpenZAP gets to see anything.

That provides a starting point for checking things.

> My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
> there is a line
> echo_cancel_level => 64
> 
> try erasing or commenting it out with a leading ;

I did a quick test, and that didn't solve the problem. I'll have to look
into it further.

> >From my recollection when you use OSLEC, you have to disable the
> dahdi/zaptel echo canceler.

That has changed in DAHDI. There is no default echo canceler. The
specific echo canceler (if any) must be specified for each channel (or
range of channels). OSLEC is referenced like any other echo canceler. For
example,

echocanceller=oslec,1-23

> BTW, you got that card for $100 because it really is worth
> that much, maybe less ;)

No arguments there. This is merely a toy system, and audio quality is not
a factor. We just want to demonstrate that this can be useful for us.
Based on past experience with Sangoma products and support, it would be
wonderful to have a card from them.

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!




From infos at madovsky.org  Fri Mar 19 11:30:17 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 14:30:17 -0400
Subject: [Freeswitch-users] rtp timeout issue
Message-ID: <30060395924C442EBEBD680E3152B9BE@MOBILEE1705>

Found the iptables lines that caused problem :

#### #Prevent external packets from using loopback
-A INPUT -i br0 -s 127.0.0.1 -j DROP
-A FORWARD -i br0 -s 127.0.0.1 -j DROP
-A INPUT -i br0 -d 127.0.0.1 -j DROP
-A FORWARD -i br0 -d 127.0.0.1 -j DROP

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:53 PM
  Subject: Re: [Freeswitch-users] rtp timeout issue


  bingo.... :), 
  grrrr, now I need to read all my iptables :(
    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Friday, March 19, 2010 1:44 PM
    Subject: Re: [Freeswitch-users] rtp timeout issue


    try stopping iptables for 1 call to make sure it is not related to that.




    On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos at madovsky.org> wrote:

      yes I did it also.
      the issue is only noticeable if I use 2 softphones on the same public IP where there is FS.
      I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost
        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Friday, March 19, 2010 1:17 PM
        Subject: Re: [Freeswitch-users] rtp timeout issue


        you should also unblock the rtp port range
        16384-32768 udp



        On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos at madovsky.org> wrote:

          with bypass-media=true everything is ok, 
          but I need to pass media through FS, I missed
          some settings but don't know where...
          I use default configs, inverted the internal and external port
          and commented out all nat params.

          Thanks
            ----- Original Message ----- 
            From: Madovsky 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Friday, March 19, 2010 12:29 PM
            Subject: rtp timeout issue


            I set FS internal on port 5080 and external 5060.
            2 softphones on public IP, so no nat.
            the sip invite is ok, but there's a media timeout..
            port 5080 and 5060 are opened on my firewall
            must I set any nat params anyway ?

            Thanks

            Franck

          _______________________________________________
          FreeSWITCH-users mailing list
          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          http://www.freeswitch.org





        -- 
        Anthony Minessale II

        FreeSWITCH http://www.freeswitch.org/
        ClueCon http://www.cluecon.com/
        Twitter: http://twitter.com/FreeSWITCH_wire

        AIM: anthm
        MSN:anthony_minessale at hotmail.com
        GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
        IRC: irc.freenode.net #freeswitch

        FreeSWITCH Developer Conference
        sip:888 at conference.freeswitch.org
        iax:guest at conference.freeswitch.org/888
        googletalk:conf+888 at conference.freeswitch.org
        pstn:+19193869900



------------------------------------------------------------------------


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org





    -- 
    Anthony Minessale II

    FreeSWITCH http://www.freeswitch.org/
    ClueCon http://www.cluecon.com/
    Twitter: http://twitter.com/FreeSWITCH_wire

    AIM: anthm
    MSN:anthony_minessale at hotmail.com
    GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
    IRC: irc.freenode.net #freeswitch

    FreeSWITCH Developer Conference
    sip:888 at conference.freeswitch.org
    iax:guest at conference.freeswitch.org/888
    googletalk:conf+888 at conference.freeswitch.org
    pstn:+19193869900



----------------------------------------------------------------------------


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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From robert.hadley at teotech.com  Fri Mar 19 11:30:55 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 11:30:55 -0700
Subject: [Freeswitch-users] How does mod_directory get users information?
In-Reply-To: <4BA3ACEA.1040503@communicatefreely.net>
References: <E70E79E7709F4FC48A49F18ED3411A49@greyhawk.tonecommander.com>
	<4BA3ACEA.1040503@communicatefreely.net>
Message-ID: <26B2C7A0869F40CDBBA7457B31FF9B75@greyhawk.tonecommander.com>

Thanks Tim,

It's working now by setting the "effective_caller_id_name" variables to
actual names.

Question: What is the way to adjust the sound level of the directory
phrases, especially the TTS portions?

The playback of the .wav files sound loud enough, but I can barely hear to
spelled names.

Thanks,
Robert


-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net] 
Sent: Friday, March 19, 2010 9:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How does mod_directory get users
information?

It looks in the user directory.

It will request users in the domain specified when you invoke it.

Something I learned (with help from the author), was that the users have to
be in a group.  The
default group will do.

If you are using xml_curl to generate your directory dynamically, you have
to set your script up so
that it will return all users in the domain when the user isn't specified in
the query.  You don't
have to return all the variables though, just the user's name and number
will usually work.

Robert Hadley wrote:
> Hello,
> 
>  
> 
> Where does mod_directory get the user information it uses to put into
> the database in creates?
> 
>  
> 
> Thanks,
> 
> Robert
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org







From fraserredmond at gmail.com  Fri Mar 19 11:43:28 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Fri, 19 Mar 2010 18:43:28 +0000
Subject: [Freeswitch-users] Open source softphone that can handle g729
Message-ID: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>

We're looking for an open source softphones that can handle g729 that we can
extend with extra functionality (connecting to other applications, and
things on the intranet.)

Does anyone have any they'd suggest?

Brian West suggested FSComm with the qualifier "it still needs more work" -
has anyone used it much (especially in windows), or know what it's
development status is? (Looking in the source browser jmesquita spent a lot
of time on it in January, but not much since.)

Cheers,
Fraser
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From brian at freeswitch.org  Fri Mar 19 11:48:32 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 19 Mar 2010 13:48:32 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <85353439-E829-4907-8B74-F877D80BCFB3@freeswitch.org>

Someone that really needs it such as yourself might think about helping the project along with some $$ ;)

FSComm will be able to use the same G729 that FreeSWITCH uses once its officially launched.

/b

On Mar 19, 2010, at 1:43 PM, Fraser Redmond wrote:

> We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.)
> 
> Does anyone have any they'd suggest?
> 
> Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.)
> 
> Cheers,
> Fraser




From anthony.minessale at gmail.com  Fri Mar 19 12:00:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:00:27 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
Message-ID: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>

I had a look at the code and I discovered that the zt mod defaults to a echo
can level of 64 internally.
I did a patch to the code that should allow it to be explicitly off when you
omit that param.

If you could update to latest trunk and test it, it should be a real test of
my theory.

If you still have the problem the next step would be to add some debugging
code to
the ozmod_zt read and write functions to see if it's sending data up to the
app.

Maybe there are some new ioctls or something they added for interop with
OSLEC that we don't know about.



On Fri, Mar 19, 2010 at 1:21 PM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
>
> > All OpenZAP does with digium cards is open the device and read
> > and write from it. If you are getting no audio the problem is
> > happening in the kernel before OpenZAP gets to see anything.
>
> That provides a starting point for checking things.
>
> > My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
> > there is a line
> > echo_cancel_level => 64
> >
> > try erasing or commenting it out with a leading ;
>
> I did a quick test, and that didn't solve the problem. I'll have to look
> into it further.
>
> > >From my recollection when you use OSLEC, you have to disable the
> > dahdi/zaptel echo canceler.
>
> That has changed in DAHDI. There is no default echo canceler. The
> specific echo canceler (if any) must be specified for each channel (or
> range of channels). OSLEC is referenced like any other echo canceler. For
> example,
>
> echocanceller=oslec,1-23
>
> > BTW, you got that card for $100 because it really is worth
> > that much, maybe less ;)
>
> No arguments there. This is merely a toy system, and audio quality is not
> a factor. We just want to demonstrate that this can be useful for us.
> Based on past experience with Sangoma products and support, it would be
> wonderful to have a card from them.
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar 19 12:03:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:03:10 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Conference with FSComm
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF46138069F8119@nickel.mettonigroup.com>
	<3181A30B8C35AB4AA8577B78DDF46138069F81D9@nickel.mettonigroup.com>
Message-ID: <191c3a031003191203q4448d227gdfd36a0f23120243@mail.gmail.com>

uuid_transfer the uuid of the holding leg first
then uuid_transfer -both on the A leg of the bridged call to

uuid_transfer <holding uuid> conference:<conf name here>@default inline

uuid_transfer -both <a-leg uuid> conference:<conf name here>@default inline


On Fri, Mar 19, 2010 at 1:20 PM, Suneel Papineni <
Suneel.Papineni at mettoni.com> wrote:

>  I am not sure if the following method works. Please let me know how could
> I implement this through event sockets.
>
>
>
> ?A? calls FSComm and FSComm calls ?B?. Now FSComm is having call details of
> all call legs. If possible FSComm now creates a conference and join all 3
> call legs to the conference. This might solve the problem, but could you
> please let me know how can create a conference and join all the legs. Please
> let me know the commands for this. I checked with ?mod_conference? module,
> unable to find respective commands. I tried ?api conference....?, it is
> saying command not found. I tried ?conference.....? command, it is also not
> working.
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
>
>
> *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto:
> freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Suneel
> Papineni
> *Sent:* 19 March 2010 15:20
> *To:* freeswitch-dev at lists.freeswitch.org
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-dev] Conference with FSComm
>
>
>
> Hi,
>
>
>
> I am trying to achieve 3-way (or 3-party) conference scenario using FSComm.
> Scenario is as follows:
>
>
>
> An incoming call comes to FSComm from an extension ?A? and is answered. Now
> FSComm makes a call to another extension ?B? and is answered. Then I want to
> bridge all three calls and be in conference. So all three parties A, B and
> FSComm will be in conference.
>
>
>
> Could someone please give me an idea how can I achieve this scenario using
> Event Sockets. (want to test with fs_cli using commands).
>
>
>
> Thanks & Regards
>
> Suneel
>
>
>
> *************************************************************************
>
> Please consider the environment before printing this e-mail
>
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> This email and any files transmitted with it are confidential and
>
> intended solely for the use of the individual or entity to whom they
>
> are addressed. If you have received this email in error please notify
>
> the system manager.  http://www.mettoni.com
>
>
>
> Mettoni Ltd
>
> Registered in England and Wales: 4485956
>
> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
>
> *************************************************************************
>
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> Please consider the environment before printing this e-mail
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> Mettoni Ltd
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> 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN
> *************************************************************************
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar 19 12:13:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:13:24 -0500
Subject: [Freeswitch-users] Originate timeout with multiple destinations
In-Reply-To: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
References: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
Message-ID: <191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>

leg_timeout is the one for the []


On Fri, Mar 19, 2010 at 1:19 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi,
>
> I need to do an originate to multiple destinations, but I need that the
> first call has a timeout of 17 seconds, and if it's not answered, then call
> the other destinations.
>
> I was thinking something like:
>
> originate
> {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz
>
>
> I tried using the call_timeout, timeout and originate_timeout variables in
> the square brackets for the first dialstring, but it didn't work. Is there a
> way of doing this with 1 originate command?
>
>
> Thanks!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From infos at madovsky.org  Fri Mar 19 12:21:43 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 19 Mar 2010 15:21:43 -0400
Subject: [Freeswitch-users] variables in gateway profile
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
Message-ID: <E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>

Ok thanks.
but if I have the same var name with different value in 2 different gateways ?

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:58 PM
  Subject: Re: [Freeswitch-users] variables in gateway profile


  yes and you can specify which direction you want to set them on, "inbound", "outbound" or "both"

  <variables>
    <variable name="foo" data="bar" direction="inbound"/>
  </variables>



  On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to create <variable> in gateway profiles like users profiles ?

    Thanks

    Franck

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Fri Mar 19 12:49:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 14:49:15 -0500
Subject: [Freeswitch-users] variables in gateway profile
In-Reply-To: <E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
	<E8413E840CC44829A48E48F6AB1F3A79@MOBILEE1705>
Message-ID: <191c3a031003191249g10be7fbbl20023e3e8584e85b@mail.gmail.com>

yes that's ok it will get set based on which one you call


On Fri, Mar 19, 2010 at 2:21 PM, Madovsky <infos at madovsky.org> wrote:

>  Ok thanks.
> but if I have the same var name with different value in 2 different
> gateways ?
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 19, 2010 1:58 PM
> *Subject:* Re: [Freeswitch-users] variables in gateway profile
>
> yes and you can specify which direction you want to set them on, "inbound",
> "outbound" or "both"
>
> <variables>
>   <variable name="foo" data="bar" direction="inbound"/>
> </variables>
>
>
> On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Is it possible to create <variable> in gateway profiles like users
>> profiles ?
>>
>> Thanks
>>
>> Franck
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From nicolas at medularis.com  Fri Mar 19 13:05:22 2010
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 19 Mar 2010 16:05:22 -0400
Subject: [Freeswitch-users] Originate timeout with multiple destinations
In-Reply-To: <191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>
References: <1b46b4e81003191119o144fa578x7b676ea94806caef@mail.gmail.com>
	<191c3a031003191213v4df1cd51w673483375fec168c@mail.gmail.com>
Message-ID: <1b46b4e81003191305y1ea80c49j27832dd1e6a488a4@mail.gmail.com>

Great, thanks!


On Fri, Mar 19, 2010 at 3:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> leg_timeout is the one for the []
>
>
> On Fri, Mar 19, 2010 at 1:19 PM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> Hi,
>>
>> I need to do an originate to multiple destinations, but I need that the
>> first call has a timeout of 17 seconds, and if it's not answered, then call
>> the other destinations.
>>
>> I was thinking something like:
>>
>> originate
>> {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz
>>
>>
>> I tried using the call_timeout, timeout and originate_timeout variables in
>> the square brackets for the first dialstring, but it didn't work. Is there a
>> way of doing this with 1 originate command?
>>
>>
>> Thanks!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From Russell.Mosemann at cune.org  Fri Mar 19 13:32:47 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Fri, 19 Mar 2010 15:32:47 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com><20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
Message-ID: <FD3C3036FB7A431D8AB83B100CCC6531@cune.pri>

Anthony Minessale wrote:
> I had a look at the code and I discovered that the zt mod defaults to a
> echo can level of 64 internally. I did a patch to the code that should
> allow it to be explicitly off when you omit that param.

I think that worked. On the softphone side I can hear the caller, now, but they can't hear me. I'm not at my regular computer, and I can't tell if the microphone isn't working, or if it is something else. As soon as get back, I will give it a try. Thanks!

--
Russell Mosemann




From robert.hadley at teotech.com  Fri Mar 19 13:42:56 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 13:42:56 -0700
Subject: [Freeswitch-users] Increase volume of "say" phrases to match
	play-file?
Message-ID: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>

Hello,

 

Is there a way to increase the volume of the "say" phrases to match the
"play-file" phrases?

 

I am using mod_directory, the sound files from
http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/, and
the built-in phrases from /usr/local/freeswitch/conf/lang/en/dir/sounds.xml.

 

 

Thanks,

Robert

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From kenfulmer at icstechnologysolutions.com  Fri Mar 19 14:08:11 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Fri, 19 Mar 2010 16:08:11 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
	<191c3a031003191021ja1cfec8h53e421a8ac727aed@mail.gmail.com>
Message-ID: <016f01cac7a8$48f13180$dad39480$@com>

Thanks for your reply.

 

If we only want to route calls from one provider to another or from the PSTN
to an internal PBX (and vice versa), would we use the external SIP profiles
for each "trunk"? 

 

Thanks,

 

Ken Fulmer

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, March 19, 2010 12:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

 

If you plan to transcode g729 to g711 it will impact performance
dramatically but you should still get a few hundered channels up on a nice
server box.  You could contact consulting at freeswitch.org about g729 license
options.

With no g729 you could scale a nice 8 core box into a few thousand channels
pushing rtp.

Do you have a requirement to transcode g729?  you could just use g711 right
from the phone couldn't you?
 

On Fri, Mar 19, 2010 at 10:13 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

1.       We would like to transcode from g711mu internally to g729 for PSTN
calls. Is this possible? 

2.       We are using sipX as an open source PBX. We'd like to use
FreeSwitch purely as a B2BUA for connectivity to the PSTN. Our Polycom
phones use SIP REFER messages to do call transfers and our ITSPs can't
handle those. So, we've tried the "under the hood" FreeSwitch within sipX.
It works for many applications but we need to scale a unified system for
larger customer deployments. If we want to create a trunk to an ITSP and a
trunk to an internal PBX, would we use the external gateway XML files for
both? We don't plan to point any internal phones to the FreeSwitch box
(we're using sipX as an internal PBX). 

3.       If we use multiple ITSPs, can we assign a priority to each one, so
one will be chosen first? 

4.       I understand no performance metrics are posted. Can anyone share
their experiences with passing RTP through the server? Does RTP affect
performance in a big way (very open ended question, I know). We are looking
for ball park answers so we'll have some idea of scalability for our
designs. 

 

Thanks,

 

Ken Fulmer

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From will.traenkle at yahoo.com  Fri Mar 19 14:24:06 2010
From: will.traenkle at yahoo.com (William Traenkle)
Date: Fri, 19 Mar 2010 14:24:06 -0700 (PDT)
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>
	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
Message-ID: <43380.74634.qm@web57607.mail.re1.yahoo.com>

Where/how does mod_directory associate a name with an extension?  Meaning, what variable and where does it get the "last name" of the extension user?

I searched the wiki and could not find much on mod_directory and how it works. Will this information be updated soon?

Thanks,

-Will




________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com> wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number?
>
>I heard this was a feature that was implemented or going to be implemented in freeswitch.
>
>If it has been implemented, would you send me info on how to configure it?
>
>If it has not been implemented, would you state when it is planned to be implemented?
>
>I believe the module name is mod_directory, is that correct?
>
It is mod_directory. I haven't tried it yet. Please test drive it and report back.
-MC


      
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From fs-list at communicatefreely.net  Fri Mar 19 14:39:55 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 17:39:55 -0400
Subject: [Freeswitch-users] mod_voicemail modifications
Message-ID: <4BA3EF2B.5060407@communicatefreely.net>

Anybody out there know the inner workings of the voice mail module?

It looks like I may have to modify the source code to make it do what I want.  It's so close, but I
can't quite get it to work.

I need to change it so that the voicemail_greeting_number channel variable is used to set the
greeting path, but is NOT stored back in the sql database.

I would also like to change it so that choosing the option to e-mail the message does not mark it as
deleted.  I imagine I will get many angry phone calls if I put it into production this way.  The
prompt doesn't tell you that the message will be deleted if you e-mail it, and not all of my users
have e-mail set up correctly.  I would prefer messages not to be deleted unless delete is explicitly
pressed.

I'm pretty new to programming, but I would like to give it a try (unless anyone else out there knows
a better way).

Can someone point me in the right direction?  I mostly want to know where the sql gets updated when
the variable is set.  That wasn't obvious when I looked through the code.

In terms of the e-mail causing the deletion, it looks like the e-mail routine is run in-line with
the delete routine, so I'm not sure if they can be separated easily.

Thanks for any help anyone can provide.

-Tim



From fs-list at communicatefreely.net  Fri Mar 19 14:42:46 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 19 Mar 2010 17:42:46 -0400
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <43380.74634.qm@web57607.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com>	<87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
	<43380.74634.qm@web57607.mail.re1.yahoo.com>
Message-ID: <4BA3EFD6.5030506@communicatefreely.net>

I suppose I could try to post something, now that I have it mostly figured out.  I did get it
working, even with our 100% xml_curl driven directory.

Give me a few days.

-Tim

William Traenkle wrote:
> Where/how does mod_directory associate a name with an extension? 
> Meaning, what variable and where does it get the "last name" of the
> extension user?
> 
> I searched the wiki and could not find much on mod_directory and how it
> works. Will this information be updated soon?
> 
> Thanks,
> 
> -Will
> 
> ------------------------------------------------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, March 18, 2010 11:28:44 AM
> *Subject:* Re: [Freeswitch-users] dial by name directory // mod_directory?
> 
> 
> 
> On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle
> <will.traenkle at yahoo.com <mailto:will.traenkle at yahoo.com>> wrote:
> 
>     Has freeswitch implemented the "Dial by Name" feature where a caller
>     can enter the first x number of characters of an extensions first or
>     last name and have the operator either connect the user to the
>     result and/or state the result's extension number?
> 
>     I heard this was a feature that was implemented or going to be
>     implemented in freeswitch.
> 
>     If it has been implemented, would you send me info on how to
>     configure it?
> 
>     If it has not been implemented, would you state when it is planned
>     to be implemented?
> 
>     I believe the module name is mod_directory, is that correct?
> 
> It is mod_directory. I haven't tried it yet. Please test drive it and
> report back.
> -MC
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From robert.hadley at teotech.com  Fri Mar 19 15:07:48 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 15:07:48 -0700
Subject: [Freeswitch-users] mod_voicemail modifications
In-Reply-To: <4BA3EF2B.5060407@communicatefreely.net>
References: <4BA3EF2B.5060407@communicatefreely.net>
Message-ID: <2DA362E3E39C405DA8557FE7ADA570F7@greyhawk.tonecommander.com>

Hi Tim,

I wrote up a Jira bug for the email key always deleting voicemails.  Andrew
T. responded with a patch that is supposed to fix half of the problem such
that voicemails are not deleted when emailed.  However he didn't include
using the vm-keep-local-after-email setting in the user profile.  I haven't
had a chance to try this patch yet.

See http://jira.freeswitch.org/browse/MODAPP-403

I would be interested in any voicemail fixes being made too.

Regards,
Robert



-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net] 
Sent: Friday, March 19, 2010 2:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_voicemail modifications

Anybody out there know the inner workings of the voice mail module?

It looks like I may have to modify the source code to make it do what I
want.  It's so close, but I
can't quite get it to work.

I need to change it so that the voicemail_greeting_number channel variable
is used to set the
greeting path, but is NOT stored back in the sql database.

I would also like to change it so that choosing the option to e-mail the
message does not mark it as
deleted.  I imagine I will get many angry phone calls if I put it into
production this way.  The
prompt doesn't tell you that the message will be deleted if you e-mail it,
and not all of my users
have e-mail set up correctly.  I would prefer messages not to be deleted
unless delete is explicitly
pressed.

I'm pretty new to programming, but I would like to give it a try (unless
anyone else out there knows
a better way).

Can someone point me in the right direction?  I mostly want to know where
the sql gets updated when
the variable is set.  That wasn't obvious when I looked through the code.

In terms of the e-mail causing the deletion, it looks like the e-mail
routine is run in-line with
the delete routine, so I'm not sure if they can be separated easily.

Thanks for any help anyone can provide.

-Tim






From srinivas.ksvreddy at gmail.com  Fri Mar 19 15:09:22 2010
From: srinivas.ksvreddy at gmail.com (srinivasula reddy)
Date: Sat, 20 Mar 2010 03:39:22 +0530
Subject: [Freeswitch-users] registering with auth id
Message-ID: <f8af5741003191509q251ff12bt6941e118d8b7ea57@mail.gmail.com>

HI,

i have a user 1002 in freeswitch, i am trying to register 2001 with help of
1002, mean with auth id is 1002 for me,
is it possible to register in this way

-- 
Srinivasula Reddy K
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From robert.hadley at teotech.com  Fri Mar 19 15:11:16 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 19 Mar 2010 15:11:16 -0700
Subject: [Freeswitch-users] mod_directory question
In-Reply-To: <43380.74634.qm@web57607.mail.re1.yahoo.com>
References: <487344.64278.qm@web57604.mail.re1.yahoo.com><87f2f3b91003181128v1b845035hd878a197e7273937@mail.gmail.com>
	<43380.74634.qm@web57607.mail.re1.yahoo.com>
Message-ID: <051753A8EA1B44A18831F4C0AF63CD4A@greyhawk.tonecommander.com>

Hi Will,

 

One way: Set the effective_caller_id_name in the LineId.xml files in the
profiles you are using.

<include>

  <user id="1045">

    <params>

    </params>

    <variables>

      <variable name="effective_caller_id_name" value="FirstName LastName"/>

 

 

I had asked the same question earlier today.  I have attached the answer I
got.

 

Regards,

Robert

 

 

  _____  

From: William Traenkle [mailto:will.traenkle at yahoo.com] 
Sent: Friday, March 19, 2010 2:24 PM
To: freeswitch-users at lists.freeswitch.org; msc at freeswitch.org
Subject: [Freeswitch-users] mod_directory question

 

Where/how does mod_directory associate a name with an extension?  Meaning,
what variable and where does it get the "last name" of the extension user?

I searched the wiki and could not find much on mod_directory and how it
works. Will this information be updated soon?

Thanks,

-Will

 

  _____  

From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 18, 2010 11:28:44 AM
Subject: Re: [Freeswitch-users] dial by name directory // mod_directory?




On Thu, Mar 18, 2010 at 11:15 AM, William Traenkle <will.traenkle at yahoo.com>
wrote:

Has freeswitch implemented the "Dial by Name" feature where a caller can
enter the first x number of characters of an extensions first or last name
and have the operator either connect the user to the result and/or state the
result's extension number?

I heard this was a feature that was implemented or going to be implemented
in freeswitch.

If it has been implemented, would you send me info on how to configure it?

If it has not been implemented, would you state when it is planned to be
implemented?

I believe the module name is mod_directory, is that correct?

It is mod_directory. I haven't tried it yet. Please test drive it and report
back.
-MC

 

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From: "Tim St. Pierre" <fs-list at communicatefreely.net>
Subject: Re: [Freeswitch-users] How does mod_directory get users information?
Date: Fri, 19 Mar 2010 09:57:14 -0700
Size: 4755
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100319/0c681242/attachment-0002.mht 

From msc at freeswitch.org  Fri Mar 19 16:27:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Mar 2010 16:27:11 -0700
Subject: [Freeswitch-users] Increase volume of "say" phrases to match
	play-file?
In-Reply-To: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>
References: <CC5C8737684A467CBAFDB71562FFC975@greyhawk.tonecommander.com>
Message-ID: <87f2f3b91003191627u7d2ce3e6j74f2241e4a8b99f2@mail.gmail.com>

If these are all static sound files then you should just normalize the ones
that are too loud. Brian uses this to normalize our stock english sound
files:

http://svn.freeswitch.org/svn/sounds/trunk/en/us/callie/dist.sh

-MC

On Fri, Mar 19, 2010 at 1:42 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>  Hello,
>
>
>
> Is there a way to increase the volume of the ?say? phrases to match the
> ?play-file? phrases?
>
>
>
> I am using mod_directory, the sound files from
> http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/,
> and the built-in phrases from
> /usr/local/freeswitch/conf/lang/en/dir/sounds.xml.
>
>
>
>
>
> Thanks,
>
> Robert
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Fri Mar 19 17:26:31 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sat, 20 Mar 2010 11:26:31 +1100
Subject: [Freeswitch-users] dahdi problems
In-Reply-To: <c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>
References: <3c5cf5261003180056h4ef20901nb1e1629f83863b2b@mail.gmail.com>
	<20100318132319.2A5BA312875@cuneorg-email.cune.pri>
	<c4d05cbe1003180719q72b19651jbda9cd35665585fc@mail.gmail.com>
Message-ID: <3c5cf5261003191726n4acc3bc0l306e38e6f0da4bde@mail.gmail.com>

On 19 March 2010 01:19, Moises Silva <moises.silva at gmail.com> wrote:
> do svn update ( that will fix some of the errors but NOT the critical one).
> The critical one is "failure configuring device ... " and I think that one
> is because you did not run "dahdi_cfg" utility.

Looks like the main part of the error was my fault. I hadn't realized
my distribution (Voyage) was missing the user space tools for dahdi
(only had the kernel modules).

Now fixed.

I still do get some errors when reloading mod_openzap, even with the
latest subversion update of freeswitch:

2010-03-20 11:24:43.081334 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.081334 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.094327 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.094327 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.094327 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.094327 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.
2010-03-20 11:24:43.532271 [ERR] ozmod_analog.c:959 Failure Polling
event! [no matching descriptor]
2010-03-20 11:24:43.532271 [DEBUG] ozmod_analog.c:976 ANALOG thread ending.

Not sure if this is my fault or not.

Fortunately it seems to work fine regardless.
-- 
Brian May <brian at microcomaustralia.com.au>



From anthony.minessale at gmail.com  Fri Mar 19 18:36:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Mar 2010 20:36:41 -0500
Subject: [Freeswitch-users] mod_voicemail modifications
In-Reply-To: <191c3a031003191833g32da5897vdcad887008bd0dbe@mail.gmail.com>
References: <4BA3EF2B.5060407@communicatefreely.net>
	<2DA362E3E39C405DA8557FE7ADA570F7@greyhawk.tonecommander.com>
	<191c3a031003191833g32da5897vdcad887008bd0dbe@mail.gmail.com>
Message-ID: <191c3a031003191836t628ecb28u156aeddd0ddcb46b@mail.gmail.com>

I think I have said at least once.
If you have that email delete problem
Update to trunc and Issue make vm-sync

You have an out-of-date vm template in your config.
Once you have the correct template it will not even offer the email option
when you don't have email.

For the other reqs file a bounty or feature req jira for the desired
options.

On Mar 19, 2010 5:13 PM, "Robert Hadley" <robert.hadley at teotech.com> wrote:

Hi Tim,

I wrote up a Jira bug for the email key always deleting voicemails.  Andrew
T. responded with a patch that is supposed to fix half of the problem such
that voicemails are not deleted when emailed.  However he didn't include
using the vm-keep-local-after-email setting in the user profile.  I haven't
had a chance to try this patch yet.

See http://jira.freeswitch.org/browse/MODAPP-403

I would be interested in any voicemail fixes being made too.

Regards,
Robert




-----Original Message-----
From: Tim St. Pierre [mailto:fs-list at communicatefreely.net]
Sent: Fri...
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From sharad at coraltele.com  Fri Mar 19 22:41:42 2010
From: sharad at coraltele.com (sharad)
Date: Sat, 20 Mar 2010 11:11:42 +0530
Subject: [Freeswitch-users] How to find the number of live calls for a
	specific dialplan
Message-ID: <B91AFBDD83964B558D84B4D859794C1F@sharad>

Hi All

If there any way to find out the number of live calls for a specific dialplan.

Regards
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From tayeb.meftah at gmail.com  Sat Mar 20 03:27:16 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 20 Mar 2010 11:27:16 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <4BA4A304.9090401@gmail.com>

hi,
purchase the freeswitch G729 license and use it with FSComm
or develope your own freeswitch based softphone with G729 license from 
brian K West
Le 19/03/2010 19:43, Fraser Redmond a ?crit :
> We're looking for an open source softphones that can handle g729 that 
> we can extend with extra functionality (connecting to other 
> applications, and things on the intranet.)
>
> Does anyone have any they'd suggest?
>
> Brian West suggested FSComm with the qualifier "it still needs more 
> work" - has anyone used it much (especially in windows), or know what 
> it's development status is? (Looking in the source browser jmesquita 
> spent a lot of time on it in January, but not much since.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From raffaele.p.guidi at gmail.com  Sat Mar 20 06:24:04 2010
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 20 Mar 2010 14:24:04 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
Message-ID: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>

I believe sip communicator does handle g729. It has a strong community and a
plugin model (based on osgi) that makes it easily (ok, kinda...)
customizable. Of course the FSComm/givesomemoneytotheproject option is the
best one. In any case:
http://sip-communicator.org/

<http://sip-communicator.org/>Ciao,
    R

On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> We're looking for an open source softphones that can handle g729 that we
> can extend with extra functionality (connecting to other applications, and
> things on the intranet.)
>
> Does anyone have any they'd suggest?
>
> Brian West suggested FSComm with the qualifier "it still needs more work" -
> has anyone used it much (especially in windows), or know what it's
> development status is? (Looking in the source browser jmesquita spent a lot
> of time on it in January, but not much since.)
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Sat Mar 20 06:53:24 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 20 Mar 2010 21:53:24 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <4BA4D354.3080009@coppice.org>

On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> I believe sip communicator does handle g729. It has a strong community 
> and a plugin model (based on osgi) that makes it easily (ok, kinda...) 
> customizable. Of course the FSComm/givesomemoneytotheproject option is 
> the best one. In any case:
> http://sip-communicator.org/
>
> Ciao,
>     R
It seems one of their summer of code projects is to implement a G.729 
codec, so it doesn't seem like they have one.
>
> On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond 
> <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
>
>     We're looking for an open source softphones that can handle g729
>     that we can extend with extra functionality (connecting to other
>     applications, and things on the intranet.)
>
>     Does anyone have any they'd suggest?
>
>     Brian West suggested FSComm with the qualifier "it still needs
>     more work" - has anyone used it much (especially in windows), or
>     know what it's development status is? (Looking in the source
>     browser jmesquita spent a lot of time on it in January, but not
>     much since.)
>

Steve




From fraserredmond at gmail.com  Sat Mar 20 08:42:44 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 20 Mar 2010 15:42:44 +0000
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com> 
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com> 
	<4BA4D354.3080009@coppice.org>
Message-ID: <c32ee1cf1003200842u5dfc14f2m918f3477a0d5e7e3@mail.gmail.com>

Thanks, all good info.

FSComm still seems like it needs a lot more work before we could use it, by
which time we'd be settled on something else. Not that that rules out giving
money to the project :-)

I'll take a better look at sip-communicator, the other one we're looking at
is Qute Com - has anyone used that at all?

Cheers,
Fraser




On Sat, Mar 20, 2010 at 1:53 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> > I believe sip communicator does handle g729. It has a strong community
> > and a plugin model (based on osgi) that makes it easily (ok, kinda...)
> > customizable. Of course the FSComm/givesomemoneytotheproject option is
> > the best one. In any case:
> > http://sip-communicator.org/
> >
> > Ciao,
> >     R
> It seems one of their summer of code projects is to implement a G.729
> codec, so it doesn't seem like they have one.
> >
> > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
> > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
> >
> >     We're looking for an open source softphones that can handle g729
> >     that we can extend with extra functionality (connecting to other
> >     applications, and things on the intranet.)
> >
> >     Does anyone have any they'd suggest?
> >
> >     Brian West suggested FSComm with the qualifier "it still needs
> >     more work" - has anyone used it much (especially in windows), or
> >     know what it's development status is? (Looking in the source
> >     browser jmesquita spent a lot of time on it in January, but not
> >     much since.)
> >
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Mar 20 09:02:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 11:02:15 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <E83352F5-E954-421C-81DC-1F621E54DEDD@freeswitch.org>

sip communicator is crap.  Its trying to do way too much and does not a single thing very well at all.  If they have G729 its not legal to use.

/b

On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:

> I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
> http://sip-communicator.org/
> 
> Ciao,
>     R

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From brian at freeswitch.org  Sat Mar 20 09:02:37 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 11:02:37 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<4BA4D354.3080009@coppice.org>
Message-ID: <F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>

I haven't see many of the SOC projects ever produce anything useful as of late.

/b

On Mar 20, 2010, at 8:53 AM, Steve Underwood wrote:

> It seems one of their summer of code projects is to implement a G.729 
> codec, so it doesn't seem like they have one.




From anthony.minessale at gmail.com  Sat Mar 20 09:12:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 20 Mar 2010 11:12:39 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
Message-ID: <191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>

I think the problem is that fs sets the stack size for the core and new
threads to 240k so when the child is born from system() the new process
inherits the small stack size and sendmail needs more than 240k

If you are root, it has the priveledge to raise the stack size to the
default size of 8m and sets it back down when the command completes.

If you are not root you lack these privs and the stack size remains small.

This has been a long time problem how to make a solution that is cross
platform and will not break working configurations.

On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:

 I tried also with sudo and freeswitch user in sudoers with the same result,
segmentation fault.
Anyway my patch works perfectly now, even with sendmail (the only difference
without patch is the "From" is taken from the voicemail.tpl and not the user
variable)


>
> ----- Original Message -----
> From: Fran?ois Legal
> To: freeswitch-users at lists.freeswitch.org


> Sent: Friday, March 19, 2010 6:51 AM
> Subject: Re: [Freeswitch-users] voicemail options [tempora...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

...
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From infos at madovsky.org  Sat Mar 20 09:42:55 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:42:55 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com><d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<E83352F5-E954-421C-81DC-1F621E54DEDD@freeswitch.org>
Message-ID: <192F31E8CC78429BBE62051BBCDB9A81@MOBILEE1705>

There's linphone, maybe you can add G729 plugin
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:02 PM
  Subject: Re: [Freeswitch-users] Open source softphone that can handle g729


  sip communicator is crap.  Its trying to do way too much and does not a single thing very well at all.  If they have G729 its not legal to use.


  /b


  On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:


    I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
    http://sip-communicator.org/


    Ciao,
        R




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar 20 09:47:37 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:47:37 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705><e2459a682c4993206a5701120a82ed33@thom.fr.eu.org><49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
Message-ID: <F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>

Ok I undertand.
buy Why to continue to use pipe and STDIN since to use "filename" path works ?
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:12 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  I think the problem is that fs sets the stack size for the core and new threads to 240k so when the child is born from system() the new process inherits the small stack size and sendmail needs more than 240k

  If you are root, it has the priveledge to raise the stack size to the default size of 8m and sets it back down when the command completes.

  If you are not root you lack these privs and the stack size remains small.

  This has been a long time problem how to make a solution that is cross platform and will not break working configurations. 


    On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:


    I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
    Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)

      >
      > ----- Original Message -----
      > From: Fran?ois Legal
      > To: freeswitch-users at lists.freeswitch.org


      > Sent: Friday, March 19, 2010 6:51 AM
      > Subject: Re: [Freeswitch-users] voicemail options [tempora...


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org

    ...



------------------------------------------------------------------------------


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat Mar 20 09:56:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 12:56:05 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
Message-ID: <C222121AFFAD4392AB636DAB2F091DD5@MOBILEE1705>

Also

http://www.antisip.com/as/index.php

they have a little softphone that manages a lot of codec (G729 included)

Regards
  ----- Original Message ----- 
  From: Raffaele P. Guidi 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 9:24 AM
  Subject: Re: [Freeswitch-users] Open source softphone that can handle g729


  I believe sip communicator does handle g729. It has a strong community and a plugin model (based on osgi) that makes it easily (ok, kinda...) customizable. Of course the FSComm/givesomemoneytotheproject option is the best one. In any case: 
  http://sip-communicator.org/


  Ciao,
      R


  On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:

    We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.)

    Does anyone have any they'd suggest?

    Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.)

    Cheers,
    Fraser



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Sat Mar 20 09:58:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 20 Mar 2010 11:58:43 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
	<F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
Message-ID: <191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>

I am willing to change anything we have to as long as it works on all
platforms and we can give proper warnings to others.
Maybe we will do something like another param to say if your mailer app
needs a pipe or can accept the filename so we can do both
and leave the defaults to the current way.

Maybe we can also have another option to make a simple relay code internally
that can deliver the file to a mail relay if you set that option.



On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:

>  Ok I undertand.
> buy Why to continue to use pipe and STDIN since to use "filename" path
> works ?
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Saturday, March 20, 2010 12:12 PM
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
> I think the problem is that fs sets the stack size for the core and new
> threads to 240k so when the child is born from system() the new process
> inherits the small stack size and sendmail needs more than 240k
>
> If you are root, it has the priveledge to raise the stack size to the
> default size of 8m and sets it back down when the command completes.
>
> If you are not root you lack these privs and the stack size remains small.
>
> This has been a long time problem how to make a solution that is cross
> platform and will not break working configurations.
>
> On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:
>
>  I tried also with sudo and freeswitch user in sudoers with the same
> result, segmentation fault.
> Anyway my patch works perfectly now, even with sendmail (the only
> difference without patch is the "From" is taken from the voicemail.tpl and
> not the user variable)
>
>
> >
> > ----- Original Message -----
> > From: Fran?ois Legal
> > To: freeswitch-users at lists.freeswitch.org
>
>
> > Sent: Friday, March 19, 2010 6:51 AM
> > Subject: Re: [Freeswitch-users] voicemail options [tempora...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ...
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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Twitter: http://twitter.com/FreeSWITCH_wire

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MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Sat Mar 20 10:13:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 20 Mar 2010 13:13:10 -0400
Subject: [Freeswitch-users] voicemail options [temporarly solved]
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705><e2459a682c4993206a5701120a82ed33@thom.fr.eu.org><49B3842E181347948484F22AB2FABBD0@MOBILEE1705><191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com><F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
Message-ID: <A643303450404497B14F79175BEC0EF4@MOBILEE1705>

ok irght.
So for now I will continue to use my patch to make it work.
I will give to you the status of any system upgrade that resolves the problem
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 20, 2010 12:58 PM
  Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


  I am willing to change anything we have to as long as it works on all platforms and we can give proper warnings to others.
  Maybe we will do something like another param to say if your mailer app needs a pipe or can accept the filename so we can do both
  and leave the defaults to the current way.

  Maybe we can also have another option to make a simple relay code internally that can deliver the file to a mail relay if you set that option.




  On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:

    Ok I undertand.
    buy Why to continue to use pipe and STDIN since to use "filename" path works ?
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Saturday, March 20, 2010 12:12 PM
      Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]


      I think the problem is that fs sets the stack size for the core and new threads to 240k so when the child is born from system() the new process inherits the small stack size and sendmail needs more than 240k

      If you are root, it has the priveledge to raise the stack size to the default size of 8m and sets it back down when the command completes.

      If you are not root you lack these privs and the stack size remains small.

      This has been a long time problem how to make a solution that is cross platform and will not break working configurations. 


        On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:


        I tried also with sudo and freeswitch user in sudoers with the same result, segmentation fault.
        Anyway my patch works perfectly now, even with sendmail (the only difference without patch is the "From" is taken from the voicemail.tpl and not the user variable)

          >
          > ----- Original Message -----
          > From: Fran?ois Legal
          > To: freeswitch-users at lists.freeswitch.org


          > Sent: Friday, March 19, 2010 6:51 AM
          > Subject: Re: [Freeswitch-users] voicemail options [tempora...


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org

        ...



--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sean at obscuradigital.com  Sat Mar 20 12:21:25 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 20 Mar 2010 12:21:25 -0700
Subject: [Freeswitch-users] Paging
Message-ID: <C7CA6E45.7010%sean@obscuradigital.com>

Hello Dev team,

Freeswitch rocks!!

I have a question about the paging application.  I?ve followed the
instructions on the wiki on how to set it up, but when I do a sip trace I?m
noticing a Proxy authentication message when try to connect to the paging
extension.  Every time I dial the extension it hungs up. I?ve checked the
acl xml file and my subnet is in the access list.  Any thoughts on how to
correct the authentication issue, if that?s issue.

Thanks
Sean
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From mastermind202 at gmail.com  Sat Mar 20 17:01:06 2010
From: mastermind202 at gmail.com (mm_202)
Date: Sat, 20 Mar 2010 20:01:06 -0400
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CA6E45.7010%sean@obscuradigital.com>
References: <C7CA6E45.7010%sean@obscuradigital.com>
Message-ID: <63de75711003201701l49237f2dh153cc4e9ba149018@mail.gmail.com>

On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
> Hello Dev team,
>
> Freeswitch rocks!!
>
> I have a question about the paging application. ?I?ve followed the
> instructions on the wiki on how to set it up, but when I do a sip trace I?m
> noticing a Proxy authentication message when try to connect to the paging
> extension. ?Every time I dial the extension it hungs up. I?ve checked the
> acl xml file and my subnet is in the access list. ?Any thoughts on how to
> correct the authentication issue, if that?s issue.
>
> Thanks
> Sean
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

Sean,

We need more information to help you.  Please include the relevant
piece(s) of your dialplan, the paging hardware, and what version of FS
you are using.  (if you are not using trunk, please update to it to
see if that may fix your problem)

-- mm_202.



From sean at obscuradigital.com  Sat Mar 20 17:35:10 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 20 Mar 2010 17:35:10 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <63de75711003201701l49237f2dh153cc4e9ba149018@mail.gmail.com>
Message-ID: <C7CAB7CE.7017%sean@obscuradigital.com>

Hey mm_202,

Thanks for the response.  Sorry for the lack of info, just getting use to
using freeswitch and not sure what's necessary to troubleshoot all FS
issues.

Here's my dialplan entry:
 <extension name="rtp_multicast_page">
     <condition field="destination_number" expression="^pagegroup$|^7243">
       <action application="answer"/>
       <action application="esf_page_group"/>
     </condition>
   </extension>

When you say "paging hardware" are you talking about my phones? Polycom 321

I just installed FS this morning so I'm sure I have the most recent version.
FYI I'm using freepbx to configure FS.

Do I need to add a dial group to get the paging working? Still learning, but
so far everything else is working.

Sean


On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:

> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello Dev team,
>> 
>> Freeswitch rocks!!
>> 
>> I have a question about the paging application. ?I?ve followed the
>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>> noticing a Proxy authentication message when try to connect to the paging
>> extension. ?Every time I dial the extension it hungs up. I?ve checked the
>> acl xml file and my subnet is in the access list. ?Any thoughts on how to
>> correct the authentication issue, if that?s issue.
>> 
>> Thanks
>> Sean
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> Sean,
> 
> We need more information to help you.  Please include the relevant
> piece(s) of your dialplan, the paging hardware, and what version of FS
> you are using.  (if you are not using trunk, please update to it to
> see if that may fix your problem)
> 
> -- mm_202.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Sat Mar 20 17:41:12 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 19:41:12 -0500
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CAB7CE.7017%sean@obscuradigital.com>
References: <C7CAB7CE.7017%sean@obscuradigital.com>
Message-ID: <5CFB0E7E-191C-494E-8D82-BEAA80496DF1@freeswitch.org>

esf_page_group is multicast paging the polycom doesn't support it.. please see the mad boss examples in the dialplan to perform it for the polycom and just about any other phone.

/b

On Mar 20, 2010, at 7:35 PM, Sean Holt wrote:

> Hey mm_202,
> 
> Thanks for the response.  Sorry for the lack of info, just getting use to
> using freeswitch and not sure what's necessary to troubleshoot all FS
> issues.
> 
> Here's my dialplan entry:
> <extension name="rtp_multicast_page">
>     <condition field="destination_number" expression="^pagegroup$|^7243">
>       <action application="answer"/>
>       <action application="esf_page_group"/>
>     </condition>
>   </extension>
> 
> When you say "paging hardware" are you talking about my phones? Polycom 321
> 
> I just installed FS this morning so I'm sure I have the most recent version.
> FYI I'm using freepbx to configure FS.
> 
> Do I need to add a dial group to get the paging working? Still learning, but
> so far everything else is working.
> 
> Sean
> 
> 
> On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:
> 
>> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>>> Hello Dev team,
>>> 
>>> Freeswitch rocks!!
>>> 
>>> I have a question about the paging application.  I?ve followed the
>>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>>> noticing a Proxy authentication message when try to connect to the paging
>>> extension.  Every time I dial the extension it hungs up. I?ve checked the
>>> acl xml file and my subnet is in the access list.  Any thoughts on how to
>>> correct the authentication issue, if that?s issue.
>>> 
>>> Thanks
>>> Sean
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
>> Sean,
>> 
>> We need more information to help you.  Please include the relevant
>> piece(s) of your dialplan, the paging hardware, and what version of FS
>> you are using.  (if you are not using trunk, please update to it to
>> see if that may fix your problem)
>> 
>> -- mm_202.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at microcomaustralia.com.au  Sat Mar 20 17:53:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 21 Mar 2010 11:53:30 +1100
Subject: [Freeswitch-users] freeswitch and android
Message-ID: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>

Hello,

Anybody here got freeswitch working with Android?

If so, what app did you use?

I tried sipdroid which looked very good, but unfortunately it seems it
answers incoming phone calls immediately and fakes a ring tone. I
filled a bug report, but it was closed as invalid, so far without
explanation:

http://code.google.com/p/sipdroid/issues/detail?id=383

Thanks
-- 
Brian May <brian at microcomaustralia.com.au>



From brian at freeswitch.org  Sat Mar 20 18:56:03 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 20 Mar 2010 20:56:03 -0500
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
Message-ID: <7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>

http://files.freeswitch.org/windows_installer/sipdroid.apk

It has SILK support.

/b

On Mar 20, 2010, at 7:53 PM, Brian May wrote:

> Hello,
> 
> Anybody here got freeswitch working with Android?
> 
> If so, what app did you use?
> 
> I tried sipdroid which looked very good, but unfortunately it seems it
> answers incoming phone calls immediately and fakes a ring tone. I
> filled a bug report, but it was closed as invalid, so far without
> explanation:
> 
> http://code.google.com/p/sipdroid/issues/detail?id=383
> 
> Thanks
> -- 
> Brian May <brian at microcomaustralia.com.au>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From steveu at coppice.org  Sat Mar 20 19:35:28 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 21 Mar 2010 10:35:28 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
Message-ID: <4BA585F0.4050400@coppice.org>

On 03/21/2010 12:02 AM, Brian West wrote:
> I haven't see many of the SOC projects ever produce anything useful as of late.
>    
True, but since they keep choosing the same mentoring organisations each 
year, I assume never producing anything useful is a key part of Google's 
game plan with SoC.

Steve




From yehavi.bourvine at gmail.com  Sat Mar 20 22:54:46 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 21 Mar 2010 07:54:46 +0200
Subject: [Freeswitch-users] How to upgrade configuration files when
	upgrading FreeSwitch
Message-ID: <d361ca7b1003202254i4df16131x9b5cac6e10851750@mail.gmail.com>

Hello,

  Once a week I upgrade our production system by retreiving
freeswitch-1.0.5-latest.tar.gz, building and installing it. This updates all
the binaries, but I am left with the current config files (which is good, as
some of them contains locally edited data). The problem is how to detect
(and merge) changes in the configuration files, like new parameters. I can
do manual diffs, but it is a time consuming task.

What is the BCP of doing such an upgrade?

                      Thanks! __Yehavi:
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From oseslija at gmail.com  Sun Mar 21 02:47:22 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 21 Mar 2010 10:47:22 +0100
Subject: [Freeswitch-users] openzap no audio?
In-Reply-To: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
References: <82dae1c01003172207q7c5aa277g1016601cdfd4d639@mail.gmail.com>
Message-ID: <4468a6771003210247r6e74246cqcd0d7e6341f5d86@mail.gmail.com>

I had the same issues with Dahdi. When I was using latest Zaptel, PRI-PRI
calls had audio.

Ognjen

On Thu, Mar 18, 2010 at 6:07 AM, Tom Christensen <pavera at gmail.com> wrote:

> I've got freeswitch installed with a digium te110p hooked up to a PRI.
>
> calls through the system work fine SIP->SIP, calling and receiving calls
> over the PRI works, (IE the signalling part, a call can be made and incoming
> calls come in over the PRI and I can route them based on DID), however,
> there is no audio to either party.  I've checked, and there is an RTP stream
> running while the call is up, there is no reason it would be blocked (IE,
> SIP->SIP calls use the same RTP port range, and they work fine).  I saw a
> few posts that talked about no audio for a few seconds after a call is
> connected, or that calls would connect and then drop after a few seconds.
>  This is different, I can leave the call up for many minutes, it doesn't
> drop, and audio never starts coming through.
>
> I'm running openzap natively, no libpri.  I am using the dahdi 2.2.1, tried
> the latest release of zaptel first, freeswitch wouldn't even load with that.
>
> I'm going to try libpri now I think as I've exhausted all the other options
> I can think of.
>
> Any input/ideas would be greatly appreciated.
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ben at langfeld.co.uk  Sun Mar 21 07:42:29 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 21 Mar 2010 14:42:29 +0000
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
	<191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
Message-ID: <fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>

Anthony,

Apologies for the delay in responding here - i've only just got around to
testing this more. After finding a 404 response to BYE, I appear to have
managed to fix this by changing the port the Line1 component of the ATA runs
on to something other than 5060. Looks like everything is working now.

Regards,
Ben Langfeld

Wave > Email


On Fri, Feb 26, 2010 at 8:01 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you would have to be more specific.
> We would need to have you test this on latest SVN trunk with a full debug
> log.
>
> enter the following at your cli and reproduce the call saving all the
> output.
>
> console loglevel debug
> sofia profile internal siptrace on
>
>
>
> On Fri, Feb 26, 2010 at 12:41 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> Hey,
>>
>> I have a small 7 seat mostly softphone based PBX installation in place
>> using freeswitch and a couple of SPA3000s for PSTN termination. Currently,
>> aside from PSTN HUP detection, outbound calls are fine setup to directly
>> dial the SPAs. When the internal VoIP side hangs up, freeswitch sends
>> NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely.
>>
>> Unfortunately inbound calls aren't so successful. Calls are sent using the
>> SPA dialplan to an internal freeswitch extension, and the calls connect
>> fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to
>> the ATA. Is there a reason for this?
>>
>> Can anyone give me any idea how I can get freeswitch to instruct the SPA
>> to drop the line?
>>
>> Regards,
>> Ben Langfeld
>>
>> Wave > Email
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From xengelpublicx at gmail.com  Sun Mar 21 07:59:20 2010
From: xengelpublicx at gmail.com (Vladimir Elizarov)
Date: Sun, 21 Mar 2010 17:59:20 +0300
Subject: [Freeswitch-users] not registered external softphone's with
	internal profile
Message-ID: <eb0385731003210759j5698f140g88061cf3c66ffc84@mail.gmail.com>

Hello,

I'm trying to register on the fs external softphone (with public ip).
Copied profine internal to internal5090
(http://pastebin.freeswitch.org/12506). When you register get 405
Method Not Allowed or 481 Subscription Does Not Exist
(http://pastebin.freeswitch.org/12507 and
http://pastebin.freeswitch.org/12508). Depending on what softphone
use. sofia_contact shows that the username is not registered
(error/user_not_registered), although it is in sofia status
internal5090 (http://pastebin.freeswitch.org/12509). What causes the
appearance of 405, 481 and the absence of registration?

-- 
Best regards, Vladimir Elizarov



From sean at obscuradigital.com  Sun Mar 21 09:35:14 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sun, 21 Mar 2010 09:35:14 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <5CFB0E7E-191C-494E-8D82-BEAA80496DF1@freeswitch.org>
Message-ID: <C7CB98D2.702F%sean@obscuradigital.com>

Thanks Brian, I'll check out mad_boss and see if it works for my setup.

Also I wanted to point in the wiki documentation for valet_park there's a
potential typo.

 <condition field="destination_number" expression="$(6000)^">

My basic understanding is the $ sign should be at the end not in the front?
I couldn't get this to work until I switched the ^ and $ around.

Sean


On 3/20/10 5:41 PM, "Brian West" <brian at freeswitch.org> wrote:

> esf_page_group is multicast paging the polycom doesn't support it.. please see
> the mad boss examples in the dialplan to perform it for the polycom and just
> about any other phone.
> 
> /b
> 
> On Mar 20, 2010, at 7:35 PM, Sean Holt wrote:
> 
>> Hey mm_202,
>> 
>> Thanks for the response.  Sorry for the lack of info, just getting use to
>> using freeswitch and not sure what's necessary to troubleshoot all FS
>> issues.
>> 
>> Here's my dialplan entry:
>> <extension name="rtp_multicast_page">
>>     <condition field="destination_number" expression="^pagegroup$|^7243">
>>       <action application="answer"/>
>>       <action application="esf_page_group"/>
>>     </condition>
>>   </extension>
>> 
>> When you say "paging hardware" are you talking about my phones? Polycom 321
>> 
>> I just installed FS this morning so I'm sure I have the most recent version.
>> FYI I'm using freepbx to configure FS.
>> 
>> Do I need to add a dial group to get the paging working? Still learning, but
>> so far everything else is working.
>> 
>> Sean
>> 
>> 
>> On 3/20/10 5:01 PM, "mm_202" <mastermind202 at gmail.com> wrote:
>> 
>>> On Sat, Mar 20, 2010 at 3:21 PM, Sean Holt <sean at obscuradigital.com> wrote:
>>>> Hello Dev team,
>>>> 
>>>> Freeswitch rocks!!
>>>> 
>>>> I have a question about the paging application.  I?ve followed the
>>>> instructions on the wiki on how to set it up, but when I do a sip trace I?m
>>>> noticing a Proxy authentication message when try to connect to the paging
>>>> extension.  Every time I dial the extension it hungs up. I?ve checked the
>>>> acl xml file and my subnet is in the access list.  Any thoughts on how to
>>>> correct the authentication issue, if that?s issue.
>>>> 
>>>> Thanks
>>>> Sean
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> Sean,
>>> 
>>> We need more information to help you.  Please include the relevant
>>> piece(s) of your dialplan, the paging hardware, and what version of FS
>>> you are using.  (if you are not using trunk, please update to it to
>>> see if that may fix your problem)
>>> 
>>> -- mm_202.
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Sun Mar 21 09:56:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 11:56:34 -0500
Subject: [Freeswitch-users] Paging
In-Reply-To: <C7CB98D2.702F%sean@obscuradigital.com>
References: <C7CB98D2.702F%sean@obscuradigital.com>
Message-ID: <FE269726-9DC4-41BC-85B4-6763AED888C5@freeswitch.org>

Its a wiki.. correct it yourself if you have a moment.

/b

On Mar 21, 2010, at 11:35 AM, Sean Holt wrote:

> Also I wanted to point in the wiki documentation for valet_park there's a
> potential typo.




From brian at freeswitch.org  Sun Mar 21 10:07:39 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 12:07:39 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA585F0.4050400@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<F11759B6-8C7D-4AD9-91F2-45D7CE8AE7FB@freeswitch.org>
	<4BA585F0.4050400@coppice.org>
Message-ID: <8D3D830E-319A-4A0D-9CC2-772B3ACC1B00@freeswitch.org>

We have applied twice now and didn't get picked either time.

/b

On Mar 20, 2010, at 9:35 PM, Steve Underwood wrote:

> True, but since they keep choosing the same mentoring organisations each 
> year, I assume never producing anything useful is a key part of Google's 
> game plan with SoC.
> 
> Steve




From sean at obscuradigital.com  Sun Mar 21 10:10:40 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sun, 21 Mar 2010 10:10:40 -0700
Subject: [Freeswitch-users] Paging
In-Reply-To: <FE269726-9DC4-41BC-85B4-6763AED888C5@freeswitch.org>
Message-ID: <C7CBA120.7032%sean@obscuradigital.com>

Didn't know I had that kind of access, but I will gladly.

Sean


On 3/21/10 9:56 AM, "Brian West" <brian at freeswitch.org> wrote:

> Its a wiki.. correct it yourself if you have a moment.
> 
> /b
> 
> On Mar 21, 2010, at 11:35 AM, Sean Holt wrote:
> 
>> Also I wanted to point in the wiki documentation for valet_park there's a
>> potential typo.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From brian at freeswitch.org  Sun Mar 21 10:24:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 12:24:52 -0500
Subject: [Freeswitch-users] Freeswitch SPA3000 HUP Not Sent
In-Reply-To: <fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>
References: <fb4ab1e41002261041v381dd63ejafac36712f33cf96@mail.gmail.com>
	<191c3a031002261201s40a7a53cyfbe30529b77918eb@mail.gmail.com>
	<fb4ab1e41003210742q49eafa19u9cbd8cdf3110a83b@mail.gmail.com>
Message-ID: <91A721BA-A79D-4715-AD76-DD8616AE8DC4@freeswitch.org>

Sounds like you had a SIP ALG in the way.

/b

On Mar 21, 2010, at 9:42 AM, Ben Langfeld wrote:

> Anthony,
> 
> Apologies for the delay in responding here - i've only just got around to testing this more. After finding a 404 response to BYE, I appear to have managed to fix this by changing the port the Line1 component of the ATA runs on to something other than 5060. Looks like everything is working now.
> 
> Regards,
> Ben Langfeld
> 
> Wave > Email




From raffaele.p.guidi at gmail.com  Sun Mar 21 11:35:31 2010
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 21 Mar 2010 19:35:31 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA4D354.3080009@coppice.org>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>
	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>
	<4BA4D354.3080009@coppice.org>
Message-ID: <d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>

Uhm, I've seen this package (/sip/communicator/impl/media/codec/audio/g729*)
dated back in 2009-06-27 *

*
http://dev.sip-communicator.org/stats/dir_src_net_java_sip_communicator_impl_media_codec_audio_g729.html
*
*
*
*with some 10 thousand lines of code and thought it was there but I'm never
tried that. I also second Brian thoughts about sip comm guys trying to do
too much. But the thing kinda works and even they have not reached any
stable build the phone is usable. Also it is in java - and this helps in
enterprises where C is a bit forgotten. Another good one in this sense was
Sipek 2 (in C#) based on pjsip (I've done real business with that, a sip
enabled browser for a small call center) but too bad development is idle
since last year.*

On Sat, Mar 20, 2010 at 2:53 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
> > I believe sip communicator does handle g729. It has a strong community
> > and a plugin model (based on osgi) that makes it easily (ok, kinda...)
> > customizable. Of course the FSComm/givesomemoneytotheproject option is
> > the best one. In any case:
> > http://sip-communicator.org/
> >
> > Ciao,
> >     R
> It seems one of their summer of code projects is to implement a G.729
> codec, so it doesn't seem like they have one.
> >
> > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
> > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>> wrote:
> >
> >     We're looking for an open source softphones that can handle g729
> >     that we can extend with extra functionality (connecting to other
> >     applications, and things on the intranet.)
> >
> >     Does anyone have any they'd suggest?
> >
> >     Brian West suggested FSComm with the qualifier "it still needs
> >     more work" - has anyone used it much (especially in windows), or
> >     know what it's development status is? (Looking in the source
> >     browser jmesquita spent a lot of time on it in January, but not
> >     much since.)
> >
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Mar 21 12:09:33 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 15:09:33 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>

last trunk,

remove current conf folder and reinstalled the default one after make install.
I stopped iptables, connect 1000 to 1001 on port 5060, but
after rtp-timeout it closes yet.

sip trace after the rtp timeout
SIP/2.0 481 Call Does Not Exist


I start to be very frustrated....
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From freeswitch-users-list at metik.com  Sun Mar 21 12:46:43 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Sun, 21 Mar 2010 15:46:43 -0400
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>
References: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>
Message-ID: <4BA677A3.4050407@metik.com>

I haven't followed this particular thread very closely but are you sure 
there are no port conflicts exhibited in the SDP?  This is UDP after all 
and unlike TCP, binding is exclusive and port selection would then need 
to take in account those ports already in use (the endpoint and FS would 
be pretty much flying blind if the endpoint is not a local process).

I would suggest reviewing your endpoint configurations and making sure 
that they are using unique (rtp/media) port ranges.  One would think 
that if the endpoint shares the same global IP address as used by FS and 
it is not a (softphone) process running on the same physical box, then 
it would be communicating with FS using its private IP. 

Does both the signaling and media traffic traverse (i.e. routed) through 
a broadband router that sits in front of both FS and the endpoint or is 
it sent (i.e. forwarded) directly?

-metik

Madovsky wrote:
> last trunk,
>  
> remove current conf folder and reinstalled the default one after make 
> install.
> I stopped iptables, connect 1000 to 1001 on port 5060, but
> after rtp-timeout it closes yet.
>  
> sip trace after the rtp timeout
> SIP/2.0 481 Call Does Not Exist
>  
> I start to be very frustrated....
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From infos at madovsky.org  Sun Mar 21 12:51:04 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 15:51:04 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <4228F02C0FB14F8684B8F057B2BC07FC@MOBILEE1705>

Apparently there s a timeout everytime
I register a phone on a public IP.
for phone behind a firewall it works well.

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 21, 2010 3:09 PM
  Subject: rtp timeout yet


  last trunk,

  remove current conf folder and reinstalled the default one after make install.
  I stopped iptables, connect 1000 to 1001 on port 5060, but
  after rtp-timeout it closes yet.

  sip trace after the rtp timeout
  SIP/2.0 481 Call Does Not Exist


  I start to be very frustrated....
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From justin at ejtown.org  Fri Mar 19 12:17:09 2010
From: justin at ejtown.org (Justin B Newman)
Date: Fri, 19 Mar 2010 15:17:09 -0400
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <00a701cac776$bf147de0$3d3d79a0$@com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
Message-ID: <27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>

> larger customer deployments. If we want to create a trunk to an ITSP and a
> trunk to an internal PBX, would we use the external gateway XML files for
> both? We don?t plan to point any internal phones to the FreeSwitch box
> (we?re using sipX as an internal PBX).

The "internal" and "external" monikers are arbitrary names given to
two SIP profiles.  It's completely possible to run a FreeSWITCH box
with a single profile, named internal, external, or general, or
whatever.  The samples are provided and so-labelled because one of the
biggest problems new users have with PBXs is distinguishing inbound
and outbound calls.  If an inbound call somehow gets into an outbound
dialplan, you can end up with a situation where someone starts using
your PBX to make calls, without your consent.

FreeSWITCH, by permitting association of ports and/or IP addresses
with specific SIP profiles makes it trivial to keep these calls
completely segregated, making it a little harder to do "stupid"
things.

-jbn



From shaikbashaatc at yahoo.com  Sat Mar 20 01:18:09 2010
From: shaikbashaatc at yahoo.com (Shaik basha)
Date: Sat, 20 Mar 2010 01:18:09 -0700 (PDT)
Subject: [Freeswitch-users] Downloading Free Switch
Message-ID: <473732.22231.qm@web43406.mail.sp1.yahoo.com>

Dear All,

Good Afternoon every one in this forum. I am new to FS. I tried to understand about FS, but could not understand completely. Hence would any one help me from where I can download FS and what FS can do exactly. 

Your early response in this regard would be very much appreciated. Thanking you in advance. Regards,

shaikh.bashaa




      




From sashker at sashker.ru  Sat Mar 20 21:31:12 2010
From: sashker at sashker.ru (=?KOI8-R?B?4czFy9PBzsTSIPzE1cHSxM/Xyd4=?=)
Date: Sun, 21 Mar 2010 09:31:12 +0500
Subject: [Freeswitch-users] I need help about my gateway
Message-ID: <aa322b471003202131y6c526f30l9f6d1d9b84652942@mail.gmail.com>

Hello!

I tried to search information about my problem at the internet and local
sites/forums, but I didn't find solution.

My provider need in CONTACT field: "7455 at providerserver.com" (7455 - is
login on provider server), but FS send:
gw+gwname at mydomain.com<gw%2Bgwname at mydomain.com>
".
If I use option "<param name="extension-in-contact" value="true"/>" in my
gateway config - FS send this: "7455 at mydomain.com".

As a result, my external calls didn't send over this provider:

"2010-03-20 17:56:30.852222 [NOTICE] switch_channel.c:669 New Channel
sofia/external/7451 [0ab674ec-bc8c-48e0-9235-
eb395dfe02b0]
2010-03-20 17:56:30.915402 [NOTICE] sofia.c:4788 Hangup sofia/external/7451
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1179 Session 28
(sofia/external/7451) Ended
2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/external/7451 [CS_DESTROY]
2010-03-20 17:56:30.919604 [INFO] mod_dptools.c:2353 Originate Failed.
Cause: NO_ROUTE_DESTINATION
2010-03-20 17:56:30.919604 [NOTICE] mod_dptools.c:2416 Hangup
sofia/internal/1000 at mydomain.com [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1179 Session 27
(sofia/internal/1000 at mydomain.com) Ended
2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1181 Close Channel
sofia/internal/1000 at mydomain.com [CS_DESTROY]"

My gateway config:

"<include>
  <gateway name="gwname">
     <param name="username" value="7455"/>
      <param name="realm" value="providerserver.com"/>
     <param name="from-user" value="7455"/>
      <param name="from-domain" value="providerserver.com"/>
      <param name="password" value="pass"/>
      <param name="extension" value="7455"/>
      <param name="proxy" value="providerserver.com"/>
      <param name="register-proxy" value="providerserver.com"/>
       <param name="expire-seconds" value="160"/>
 <param name="register" value="true"/>
     <param name="register-transport" value="udp"/>
     <param name="retry-seconds" value="30"/>
    <param name="extension-in-contact" value="true"/>
      <param name="caller-id-in-from" value="true"/>
      <param name="contact-params" value="tport=tcp"/>
      <param name="ping" value="25"/>
  </gateway>
</include>

This gateway status in "sofia status": REGED.

Sorry for my bad english )))

Best regards, Alexander
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From emcho at sip-communicator.org  Sun Mar 21 03:05:46 2010
From: emcho at sip-communicator.org (Emil Ivov)
Date: Sun, 21 Mar 2010 11:05:46 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
Message-ID: <4BA5EF7A.9010200@sip-communicator.org>

> sip communicator is crap. Its trying to do way too much and does not
> a single thing very well at all.  

Well, that's a bit vague. Maybe if you've had/have specific issues with
it, you could drop us a note so that we could fix them.

In any case the project has improved a lot during the last years, and
even months so you might want to give it another try one of these days.

> If they have G729 its not legal to use.

Yes we do have it, and it was implemented as a part of our participation
in GSoC 2009. People can use it perfectly legally as long as they buy
the corresponding licenses (just as with any other client).

Currently use of the codec is disabled by default and whoever needs it
has to enable it in the code, after acquiring the necessary license.

Cheers,
Emil

> 
> /b
> 
> On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:
> 
>> I believe sip communicator does handle g729. It has a strong
>> community and a plugin model (based on osgi) that makes it easily
>> (ok, kinda...) customizable. Of course the
>> FSComm/givesomemoneytotheproject option is the best one. In any
>> case: http://sip-communicator.org/
>> 
>> Ciao, R
> 




From edpimentl at gmail.com  Sun Mar 21 16:41:12 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Sun, 21 Mar 2010 19:41:12 -0400
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA5EF7A.9010200@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
Message-ID: <9dc4a1671003211641q3cc1a088w928cbccea4c1443b@mail.gmail.com>

Here are two softphone supporting g729
http://www.flaphone.com/help.php#about
http://trac.qutecom.org/wiki/SupportedCodecs

Also the above softphone  may provide good UI ideas for FSComm
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From steveu at coppice.org  Sun Mar 21 16:43:43 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 22 Mar 2010 07:43:43 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>
References: <c32ee1cf1003191143u4dd93d56yc0eb5a95954c77ce@mail.gmail.com>	<d5828a511003200624v16356482q906c7e7b526eb1f0@mail.gmail.com>	<4BA4D354.3080009@coppice.org>
	<d5828a511003211135n601ff2c4lc103f9b83ad19fa7@mail.gmail.com>
Message-ID: <4BA6AF2F.9090909@coppice.org>

On 03/22/2010 02:35 AM, Raffaele P. Guidi wrote:
> Uhm, I've seen this package 
> (/sip/communicator/impl/media/codec/audio/g729*) dated back 
> in 2009-06-27* **
>
> **http://dev.sip-communicator.org/stats/dir_src_net_java_sip_communicator_impl_media_codec_audio_g729.html
> **
> **
> **
> **with some 10 thousand lines of code and thought it was there but I'm 
> never tried that. I also second Brian thoughts about sip comm guys 
> trying to do too much. But the thing kinda works and even they have 
> not reached any stable build the phone is usable. Also it is in java - 
> and this helps in enterprises where C is a bit forgotten. Another good 
> one in this sense was Sipek 2 (in C#) based on pjsip (I've done real 
> business with that, a sip enabled browser for a small call center) but 
> too bad development is idle since last year.**
That does look like the complete ITU reference codec translated into 
Java. Maybe the SoC project is to move it forwrad, rather than start it. 
The SoC description certainly looks like its to start it. I believe its 
really hard to lock down and licence modules of Java code. I wonder if 
the even have a plan for how this could be legally used.

> On Sat, Mar 20, 2010 at 2:53 PM, Steve Underwood <steveu at coppice.org 
> <mailto:steveu at coppice.org>> wrote:
>
>     On 03/20/2010 09:24 PM, Raffaele P. Guidi wrote:
>     > I believe sip communicator does handle g729. It has a strong
>     community
>     > and a plugin model (based on osgi) that makes it easily (ok,
>     kinda...)
>     > customizable. Of course the FSComm/givesomemoneytotheproject
>     option is
>     > the best one. In any case:
>     > http://sip-communicator.org/
>     >
>     > Ciao,
>     >     R
>     It seems one of their summer of code projects is to implement a G.729
>     codec, so it doesn't seem like they have one.
>     >
>     > On Fri, Mar 19, 2010 at 7:43 PM, Fraser Redmond
>     > <fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>
>     <mailto:fraserredmond at gmail.com <mailto:fraserredmond at gmail.com>>>
>     wrote:
>     >
>     >     We're looking for an open source softphones that can handle g729
>     >     that we can extend with extra functionality (connecting to other
>     >     applications, and things on the intranet.)
>     >
>     >     Does anyone have any they'd suggest?
>     >
>     >     Brian West suggested FSComm with the qualifier "it still needs
>     >     more work" - has anyone used it much (especially in windows), or
>     >     know what it's development status is? (Looking in the source
>     >     browser jmesquita spent a lot of time on it in January, but not
>     >     much since.)
>     >
>
Steve




From brian at freeswitch.org  Sun Mar 21 16:57:15 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 18:57:15 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA5EF7A.9010200@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
Message-ID: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>


On Mar 21, 2010, at 5:05 AM, Emil Ivov wrote:

>> sip communicator is crap. Its trying to do way too much and does not
>> a single thing very well at all.  
> 
> Well, that's a bit vague. Maybe if you've had/have specific issues with
> it, you could drop us a note so that we could fix them.

Overall UI behavior is not well tested.  Too many options, too many clicks to actually get to what you want to do.  Video with H264 doesn't work so well from SIP-Communicator to SIP-Communicator as recent as last weeks build.  I'm pretty sure my expectations are higher then most people's as I find bugs in everything I get my hands on.  I usually give up at around two or three bugs.  I understand you're open source as are we and thats hard work.  I'm fairly critical of everything so you can ignore me most of the time.  

> In any case the project has improved a lot during the last years, and
> even months so you might want to give it another try one of these days.

See comment above, Not sure what was going on or what was wrong... it just didn't work as expected or like EyeBeam or X-lite does.

>> If they have G729 its not legal to use.
> 
> Yes we do have it, and it was implemented as a part of our participation
> in GSoC 2009. People can use it perfectly legally as long as they buy
> the corresponding licenses (just as with any other client).

You're facilitating infringement.  Nobody can buy a license legally for a hand full of channels. 

> Currently use of the codec is disabled by default and whoever needs it
> has to enable it in the code, after acquiring the necessary license.

Like they really go get a license, where are they going to do this for one channel?

> Cheers,
> Emil




From brian at microcomaustralia.com.au  Sun Mar 21 17:10:50 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Mar 2010 11:10:50 +1100
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
	<7402939A-CBC3-4E4E-BA2C-00365916DA88@freeswitch.org>
Message-ID: <3c5cf5261003211710m2692352bh86f584c067d288d@mail.gmail.com>

On 21 March 2010 12:56, Brian West <brian at freeswitch.org> wrote:
> http://files.freeswitch.org/windows_installer/sipdroid.apk

sipdroid seems to be the application I am having problems with.

Unless this is a different version?

When I get a chance I will try and get a packet trace of the problem,
as the only response is that my report is invalid.
-- 
Brian May <brian at microcomaustralia.com.au>



From steveu at coppice.org  Sun Mar 21 17:23:40 2010
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 22 Mar 2010 08:23:40 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
Message-ID: <4BA6B88C.5010201@coppice.org>

On 03/22/2010 07:57 AM, Brian West wrote:
> On Mar 21, 2010, at 5:05 AM, Emil Ivov wrote:
>
>    
>>> sip communicator is crap. Its trying to do way too much and does not
>>> a single thing very well at all.
>>>        
>> Well, that's a bit vague. Maybe if you've had/have specific issues with
>> it, you could drop us a note so that we could fix them.
>>      
> Overall UI behavior is not well tested.  Too many options, too many clicks to actually get to what you want to do.  Video with H264 doesn't work so well from SIP-Communicator to SIP-Communicator as recent as last weeks build.  I'm pretty sure my expectations are higher then most people's as I find bugs in everything I get my hands on.  I usually give up at around two or three bugs.  I understand you're open source as are we and thats hard work.  I'm fairly critical of everything so you can ignore me most of the time.
>
>    
>> In any case the project has improved a lot during the last years, and
>> even months so you might want to give it another try one of these days.
>>      
> See comment above, Not sure what was going on or what was wrong... it just didn't work as expected or like EyeBeam or X-lite does.
>
>    
>>> If they have G729 its not legal to use.
>>>        
>> Yes we do have it, and it was implemented as a part of our participation
>> in GSoC 2009. People can use it perfectly legally as long as they buy
>> the corresponding licenses (just as with any other client).
>>      
> You're facilitating infringement.  Nobody can buy a license legally for a hand full of channels.
>    
Sadly, that's true. For a while you could get one off licences for the 
G.729 patent pool, though I'm not clear if the people outside the pool 
ever offered one off licences for their patents - one isn't much use 
without the other. These days, neither the patents in the pool, or the 
patents outside, can be licenced for small numbers of channels.

I know these patent pools need to be very wary of things like RICO 
charges in the US, but once a standard has become a gating point - 
G.729, H.264, V.34, V.90 etc - we really are better off with 
comprehensive patent pools, overseen by those who police corruption.
>> Currently use of the codec is disabled by default and whoever needs it
>> has to enable it in the code, after acquiring the necessary license.
>>      
> Like they really go get a license, where are they going to do this for one channel?
>
>    
Steve




From emcho at sip-communicator.org  Sun Mar 21 17:47:10 2010
From: emcho at sip-communicator.org (Emil Ivov)
Date: Mon, 22 Mar 2010 01:47:10 +0100
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
Message-ID: <4BA6BE0E.1000005@sip-communicator.org>

?? 22.03.10 00:57, Brian West ??????:
>> Well, that's a bit vague. Maybe if you've had/have specific issues
>> with it, you could drop us a note so that we could fix them.
> 
> Overall UI behavior is not well tested.  Too many options, too many
> clicks to actually get to what you want to do. 

Well, I am still not sure I see which options you have in mind so if you
ever come across them again, don't hesitate to drop us (or me) a note.

> Video with H264
> doesn't work so well from SIP-Communicator to SIP-Communicator as
> recent as last weeks build. 

Right, depending on the OS and the device, this one does have some
issues. We have developers looking at them, and expect to see them
resolved in the following couple of weeks. Could let you know when done,
if interested.

> See comment above, Not sure what was going on or what was wrong... it
> just didn't work as expected or like EyeBeam or X-lite does.

Quality-wise, I certainly don't agree. As for look and feel, the
projects are hard to compare since they are simply different
applications with pretty different objectives. This doesn't mean either
of them is ... crap.

Cheers,
Emil




From brian at freeswitch.org  Sun Mar 21 17:49:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 19:49:52 -0500
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA6BE0E.1000005@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<E1787AAB-C57E-4F36-ACBB-996ED4D89674@freeswitch.org>
	<4BA6BE0E.1000005@sip-communicator.org>
Message-ID: <CF6C2BE1-A17A-4851-A21D-50C3CD5F9D09@freeswitch.org>

Well in general the current state of Soft Phones that just work and look great is lacking.   

/b

On Mar 21, 2010, at 7:47 PM, Emil Ivov wrote:

>> See comment above, Not sure what was going on or what was wrong... it
>> just didn't work as expected or like EyeBeam or X-lite does.
> 
> Quality-wise, I certainly don't agree. As for look and feel, the
> projects are hard to compare since they are simply different
> applications with pretty different objectives. This doesn't mean either
> of them is ... crap.
> 
> Cheers,
> Emil




From dave.redmore at spigotsystems.com  Sun Mar 21 18:45:24 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Sun, 21 Mar 2010 20:45:24 -0500 (CDT)
Subject: [Freeswitch-users] Monitoring Registrations/Users
In-Reply-To: <33146668.81269222287052.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <2534666.101269222324377.JavaMail.root@zimbra1.spigotsystems.com>

Been working with Freeswitch to implement a simple T38 proxy to provide outbound faxing to some customers. Works very well, on initial testing, using the proxy-media option. Nice NAT detection/SBC functionality "out of the box". Generally very easy to get my head around and get it to do what I wanted, 

One bit of functionality I've been looking for in a softswitch for a while, and have never seen, is to somehow monitor the status of users/registrations and alert somehow if there is a problem. In my situation, we have connected some customers with VoIP phone service for their small offices. Typically, just a simple ATA that gives them 1-4 trunks and a couple of DIDs. The problem has always been that if their ATA drops it's registration or is not available, no one knows it until a customer tells them through some other means that their phones aren't working. This is a huge problem, obviously, and happens more than I would like. It seems like it wouldn't be too difficult to create some sort of monitoring function which can send an Invite packet every little while and make sure the UA responds with a 100/Trying and then disconnect. Basically polling any devices that I deem as needing to always be available and sending an email or something if one isn't. Does this seem useful to anyone else? Has anyone seen anything like this out there? 

Thanks, 

Dave Redmore 
Spigot Networks, Inc. 
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From infos at madovsky.org  Sun Mar 21 19:40:01 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 22:40:01 -0400
Subject: [Freeswitch-users] Downloading Free Switch
References: <473732.22231.qm@web43406.mail.sp1.yahoo.com>
Message-ID: <CDCE5AE6FA1B40ABBA8B03EC2B8CF10C@MOBILEE1705>

try to read wiki as thousands people did to learn FS

http://wiki.freeswitch.org/wiki/Main_Page

----- Original Message ----- 
From: "Shaik basha" <shaikbashaatc at yahoo.com>
To: <freeswitch-users at lists.freeswitch.org>; 
<freeswitch-users-request at lists.freeswitch.org>
Sent: Saturday, March 20, 2010 4:18 AM
Subject: [Freeswitch-users] Downloading Free Switch


> Dear All,
>
> Good Afternoon every one in this forum. I am new to FS. I tried to 
> understand about FS, but could not understand completely. Hence would any 
> one help me from where I can download FS and what FS can do exactly.
>
> Your early response in this regard would be very much appreciated. 
> Thanking you in advance. Regards,
>
> shaikh.bashaa
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From carlos.talbot at gmail.com  Sun Mar 21 19:39:45 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Sun, 21 Mar 2010 21:39:45 -0500
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
Message-ID: <5800526b1003211939g1a9cf076m13638514d4ebfde9@mail.gmail.com>

Brian,

I've experienced the same problem. I have several SIP clients registered to
one extension on my FreeSWITCH server at home (multiple-registrations). If
one of the clients happens to be siproid it will immediately answer while
the other clients stop ringing. It also sends a European ringback tone to
the caller for some reason. I'm still trying to debug the sipdroid code to
figure out where this is happening.

My impression is that the primary developer of sipdroid is in an Asterisk
world and so does not really care on how the mjsip based SIP stack behaves
in other environments.

Incidentally, Brian West had replied to your email regarding a copy of
sipdroid I recently built with the SILK codec. I've recently checked in the
code along with BV support. I recommend SILK over BV with sipdroid though as
BV is a CPU hog. Well, this is partly to do with how sipdroid behaves too.
See here: http://code.google.com/p/sipdroid/issues/detail?id=358

The patch includes SILK's wide band codecs so it's worth testing against
a FreeSWITCH server. I'd like others feedback on the perceived sound quality
(especially those with Droids as I hear they have a pretty decent speaker).
On my HTC magic you can tell the difference between SILK8 and 16 but I'm not
that impressed on the fidelity of the handset speaker. Most cell phone
manufactures nowadays (including Apple) don't seem to care about the primary
purpose of the device; a decent sounding phone call.

regards,

Carlos



On Sat, Mar 20, 2010 at 7:53 PM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> Anybody here got freeswitch working with Android?
>
> If so, what app did you use?
>
> I tried sipdroid which looked very good, but unfortunately it seems it
> answers incoming phone calls immediately and fakes a ring tone. I
> filled a bug report, but it was closed as invalid, so far without
> explanation:
>
> http://code.google.com/p/sipdroid/issues/detail?id=383
>
> Thanks
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun Mar 21 19:42:11 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 22:42:11 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <813E7259DBF04DC0B9CA947696D97261@MOBILEE1705>

Apparently there s a timeout everytime
I register a phone on a public IP.
for phone behind a firewall (outside FS network) it works well.

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 21, 2010 3:09 PM
  Subject: rtp timeout yet


  last trunk,

  remove current conf folder and reinstalled the default one after make install.
  I stopped iptables, connect 1000 to 1001 on port 5060, but
  after rtp-timeout it closes yet.

  sip trace after the rtp timeout
  SIP/2.0 481 Call Does Not Exist


  I start to be very frustrated....



--------------------------------------------------------------------------------


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Mar 21 20:12:51 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 23:12:51 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <4187D1079D044FFEBBFE620137D3CDEE@MOBILEE1705>
	<4BA677A3.4050407@metik.com>
Message-ID: <A094159DCE804BE1A16E84DEC4D312BE@MOBILEE1705>

Hi Metik,

Thanks for your answer.  Just wondering which rtp port FS use really.
on the sip trace I can see often rtp port up to 32768  (like 50434 etc..)
so it should be a problem since I use apache with port 50050:60100 on
the same IP as FS.


----- Original Message ----- 
From: "Metik" <freeswitch-users-list at metik.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 21, 2010 3:46 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


>I haven't followed this particular thread very closely but are you sure 
> there are no port conflicts exhibited in the SDP?  This is UDP after all 
> and unlike TCP, binding is exclusive and port selection would then need 
> to take in account those ports already in use (the endpoint and FS would 
> be pretty much flying blind if the endpoint is not a local process).
> 
> I would suggest reviewing your endpoint configurations and making sure 
> that they are using unique (rtp/media) port ranges.  One would think 
> that if the endpoint shares the same global IP address as used by FS and 
> it is not a (softphone) process running on the same physical box, then 
> it would be communicating with FS using its private IP. 
> 
> Does both the signaling and media traffic traverse (i.e. routed) through 
> a broadband router that sits in front of both FS and the endpoint or is 
> it sent (i.e. forwarded) directly?
> 
> -metik
> 
> Madovsky wrote:
>> last trunk,
>>  
>> remove current conf folder and reinstalled the default one after make 
>> install.
>> I stopped iptables, connect 1000 to 1001 on port 5060, but
>> after rtp-timeout it closes yet.
>>  
>> sip trace after the rtp timeout
>> SIP/2.0 481 Call Does Not Exist
>>  
>> I start to be very frustrated....
>>  
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Sun Mar 21 20:29:36 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 23:29:36 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>

For example,

I set rtp-start and end port in switch.conf.xml from 16000-32000
but When I register a phone it uses port above 32000.
example on a register sip trace :

Via: SIP/2.0/UDP 10.10.10.10:52641;rport=52641

Any idea ?


----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 21, 2010 11:12 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


> Hi Metik,
>
> Thanks for your answer.  Just wondering which rtp port FS use really.
> on the sip trace I can see often rtp port up to 32768  (like 50434 etc..)
> so it should be a problem since I use apache with port 50050:60100 on
> the same IP as FS.
>
>
> ----- Original Message ----- 
> From: "Metik" <freeswitch-users-list at metik.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, March 21, 2010 3:46 PM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
>>I haven't followed this particular thread very closely but are you sure 
>>there are no port conflicts exhibited in the SDP?  This is UDP after all 
>>and unlike TCP, binding is exclusive and port selection would then need to 
>>take in account those ports already in use (the endpoint and FS would be 
>>pretty much flying blind if the endpoint is not a local process).
>>
>> I would suggest reviewing your endpoint configurations and making sure 
>> that they are using unique (rtp/media) port ranges.  One would think that 
>> if the endpoint shares the same global IP address as used by FS and it is 
>> not a (softphone) process running on the same physical box, then it would 
>> be communicating with FS using its private IP. Does both the signaling 
>> and media traffic traverse (i.e. routed) through a broadband router that 
>> sits in front of both FS and the endpoint or is it sent (i.e. forwarded) 
>> directly?
>>
>> -metik
>>
>> Madovsky wrote:
>>> last trunk,
>>>  remove current conf folder and reinstalled the default one after make 
>>> install.
>>> I stopped iptables, connect 1000 to 1001 on port 5060, but
>>> after rtp-timeout it closes yet.
>>>  sip trace after the rtp timeout
>>> SIP/2.0 481 Call Does Not Exist
>>>  I start to be very frustrated....
>>>  ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 



From brian at freeswitch.org  Sun Mar 21 20:40:09 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 22:40:09 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
Message-ID: <A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org>

Really?  Are you really that silly?  RTP has nothing to do with the signaling nor does it have a single thing to do with restricting the client side ports.  Pay closer attention the rtp port range is the FreeSWITCH side of things... the client side choices can not and will not be controlled by any setting you can do in the FreeSWITCH config files.

/b

On Mar 21, 2010, at 10:29 PM, Madovsky wrote:

> For example,
> 
> I set rtp-start and end port in switch.conf.xml from 16000-32000
> but When I register a phone it uses port above 32000.
> example on a register sip trace :
> 
> Via: SIP/2.0/UDP 10.10.10.10:52641;rport=52641
> 
> Any idea ?



From infos at madovsky.org  Sun Mar 21 20:59:45 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 21 Mar 2010 23:59:45 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
	<A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org>
Message-ID: <98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>

Just to say to you, I know VOIP since 2 months....
thanks for your appreciation...

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 21, 2010 11:40 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


> Really?  Are you really that silly?  RTP has nothing to do with the 
> signaling nor does it have a single thing to do with restricting the 
> client side ports.  Pay closer attention the rtp port range is the 
> FreeSWITCH side of things... the client side choices can not and will not 
> be controlled by any setting you can do in the FreeSWITCH config files.
>
> /b
>
> On Mar 21, 2010, at 10:29 PM, Madovsky wrote:
>
>> For example,
>>
>> I set rtp-start and end port in switch.conf.xml from 16000-32000
>> but When I register a phone it uses port above 32000.
>> example on a register sip trace :
>>
>> Via: SIP/2.0/UDP 10.10.10.10:52641;rport=52641
>>
>> Any idea ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From brian at freeswitch.org  Sun Mar 21 21:10:41 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 23:10:41 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
	<A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org>
	<98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
Message-ID: <F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>

It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.

/b

On Mar 21, 2010, at 10:59 PM, Madovsky wrote:

> Just to say to you, I know VOIP since 2 months....
> thanks for your appreciation...
> 
> ----- Original Message ----- 
> From: "Brian West" <brian at freeswitch.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Sunday, March 21, 2010 11:40 PM
> Subject: Re: [Freeswitch-users] rtp timeout yet
> 
> 
>> Really?  Are you really that silly?  RTP has nothing to do with the 
>> signaling nor does it have a single thing to do with restricting the 
>> client side ports.  Pay closer attention the rtp port range is the 
>> FreeSWITCH side of things... the client side choices can not and will not 
>> be controlled by any setting you can do in the FreeSWITCH config files.
>> 
>> /b

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From infos at madovsky.org  Sun Mar 21 21:18:29 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 00:18:29 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
	<F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>
Message-ID: <3CD7D487272542BC9004A030FADB95DC@MOBILEE1705>

thanks for your advice !
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 22, 2010 12:10 AM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.


  /b


  On Mar 21, 2010, at 10:59 PM, Madovsky wrote:


    Just to say to you, I know VOIP since 2 months....
    thanks for your appreciation...

    ----- Original Message ----- 
    From: "Brian West" <brian at freeswitch.org>
    To: <freeswitch-users at lists.freeswitch.org>
    Sent: Sunday, March 21, 2010 11:40 PM
    Subject: Re: [Freeswitch-users] rtp timeout yet



      Really?  Are you really that silly?  RTP has nothing to do with the 

      signaling nor does it have a single thing to do with restricting the 

      client side ports.  Pay closer attention the rtp port range is the 

      FreeSWITCH side of things... the client side choices can not and will not 

      be controlled by any setting you can do in the FreeSWITCH config files.



      /b




------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Sun Mar 21 21:20:17 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 00:20:17 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
	<F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>
Message-ID: <DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>

So do you have an answer to the problem I explained on this thread ?
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 22, 2010 12:10 AM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.


  /b


  On Mar 21, 2010, at 10:59 PM, Madovsky wrote:


    Just to say to you, I know VOIP since 2 months....
    thanks for your appreciation...

    ----- Original Message ----- 
    From: "Brian West" <brian at freeswitch.org>
    To: <freeswitch-users at lists.freeswitch.org>
    Sent: Sunday, March 21, 2010 11:40 PM
    Subject: Re: [Freeswitch-users] rtp timeout yet



      Really?  Are you really that silly?  RTP has nothing to do with the 

      signaling nor does it have a single thing to do with restricting the 

      client side ports.  Pay closer attention the rtp port range is the 

      FreeSWITCH side of things... the client side choices can not and will not 

      be controlled by any setting you can do in the FreeSWITCH config files.



      /b




------------------------------------------------------------------------------


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Sun Mar 21 21:37:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 21 Mar 2010 23:37:07 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
	<F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>
	<DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>
Message-ID: <428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>

Without a full pcap I can only guess you have a network issue.

/b

On Mar 21, 2010, at 11:20 PM, Madovsky wrote:

> So do you have an answer to the problem I explained on this thread ?
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, March 22, 2010 12:10 AM
> Subject: Re: [Freeswitch-users] rtp timeout yet
> 
> It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.
> 
> /b

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From lloyd.aloysius at gmail.com  Sun Mar 21 22:27:28 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 22 Mar 2010 01:27:28 -0400
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
	<A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org>
	<98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
	<F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>
	<DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>
	<428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>
Message-ID: <8a19bf2e1003212227j278178afl7d9fa245ef7084e5@mail.gmail.com>

I think your router losing the state of the registration. What is the router
model?

Lloyd

On Mon, Mar 22, 2010 at 12:37 AM, Brian West <brian at freeswitch.org> wrote:

> Without a full pcap I can only guess you have a network issue.
>
> /b
>
> On Mar 21, 2010, at 11:20 PM, Madovsky wrote:
>
> So do you have an answer to the problem I explained on this thread ?
>
> ----- Original Message -----
> *From:* Brian West <brian at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, March 22, 2010 12:10 AM
> *Subject:* Re: [Freeswitch-users] rtp timeout yet
>
> It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head
> around.
>
> /b
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sun Mar 21 22:34:15 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 01:34:15 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705><F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org><DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>
	<428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>
Message-ID: <79A015FDD1C443B4B18CFA15941916DF@MOBILEE1705>

result with 

tcpdump -ennqti br0 -T rtp -vvv dst port 5060

10.10.10.132 = softphone1 registered from public IP

11.11.11.= softphone2 registered from public IP

10.10.10.138 = FS

00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 1047: (tos 0x0, ttl 122, id 61054, offset 0, flags [none], proto UDP (17), length 1033) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtpv1 997 c14 * 22089 1413816435 1768962609
00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 365: (tos 0x0, ttl 122, id 61082, offset 0, flags [none], proto UDP (17), length 351) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtpv1 315 c3 * 19232 1936289850 825241648
00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 1307: (tos 0x0, ttl 122, id 61083, offset 0, flags [none], proto UDP (17), length 1293) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtpv1 1257 c14 * 22089 1413816435 1768962609
00:04:23:b5:3f:05 > 00:e0:ed:0b:00:c2, IPv4, length 328: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 314) 10.10.10.132.59749 > 10.10.10.138.5060: udp/rtpv1 278 c9 * 20527 841887776 825241632
00:04:23:b5:3f:05 > 00:e0:ed:0b:00:c2, IPv4, length 555: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 541) 10.10.10.132.59749 > 10.10.10.138.5060: udp/rtpv1 505 c9 * 20527 841887776 842018848
00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 748: (tos 0x0, ttl 122, id 62599, offset 0, flags [none], proto UDP (17), length 734) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtpv1 698 c3 * 19232 1936289850 825241648
00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 46: (tos 0x0, ttl 122, id 62813, offset 0, flags [none], proto UDP (17), length 32) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtp -8 c10 3338 1936289850 825241648
00:04:23:b5:3f:05 > 00:e0:ed:0b:00:c2, IPv4, length 338: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 324) 10.10.10.132.59749 > 10.10.10.138.5060: udp/rtpv1 288 c9 * 20527 841887776 842018848
00:04:23:b5:3f:05 > 00:e0:ed:0b:00:c2, IPv4, length 439: (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 425) 10.10.10.132.59749 > 10.10.10.138.5060: udp/rtpv1 389 c25 * 17696 1936289850 1836016735
00:d0:04:01:ec:00 > 00:e0:ed:0b:00:c2, IPv4, length 415: (tos 0x0, ttl 122, id 63102, offset 0, flags [none], proto UDP (17), length 401) 11.11.11.38.11720 > 10.10.10.138.5060: udp/rtpv1 365 c9 * 20527 841887776 842018848



Danke












  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 22, 2010 12:37 AM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  Without a full pcap I can only guess you have a network issue.


  /b


  On Mar 21, 2010, at 11:20 PM, Madovsky wrote:


    So do you have an answer to the problem I explained on this thread ?
      ----- Original Message -----
      From: Brian West
      To: freeswitch-users at lists.freeswitch.org
      Sent: Monday, March 22, 2010 12:10 AM
      Subject: Re: [Freeswitch-users] rtp timeout yet


      It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.


      /b




------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Sun Mar 21 22:35:37 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 01:35:37 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705><F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org><DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705><428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>
	<8a19bf2e1003212227j278178afl7d9fa245ef7084e5@mail.gmail.com>
Message-ID: <988A7D6805CC4B569FD45E10E5A32538@MOBILEE1705>

no router ;)

I tried with 2 softphone directly from public IP
  ----- Original Message ----- 
  From: Aloysius Lloyd 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 22, 2010 1:27 AM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  I think your router losing the state of the registration. What is the router model?

  Lloyd


  On Mon, Mar 22, 2010 at 12:37 AM, Brian West <brian at freeswitch.org> wrote:

    Without a full pcap I can only guess you have a network issue.


    /b


    On Mar 21, 2010, at 11:20 PM, Madovsky wrote:


      So do you have an answer to the problem I explained on this thread ?
        ----- Original Message -----
        From: Brian West
        To: freeswitch-users at lists.freeswitch.org
        Sent: Monday, March 22, 2010 12:10 AM
        Subject: Re: [Freeswitch-users] rtp timeout yet


        It shows!  ;)  VoIP / Telephony isn't an easy thing to wrap your head around.


        /b



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jason at jasonjgw.net  Sun Mar 21 22:49:20 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 22 Mar 2010 16:49:20 +1100
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <79A015FDD1C443B4B18CFA15941916DF@MOBILEE1705>
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705>
	<A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org>
	<98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705>
	<F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org>
	<DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705>
	<428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org>
	<79A015FDD1C443B4B18CFA15941916DF@MOBILEE1705>
Message-ID: <20100322054920.GB10155@jdc.jasonjgw.net>

Madovsky <infos at madovsky.org> wrote:
> result with 
> 
> tcpdump -ennqti br0 -T rtp -vvv dst port 5060
> 
> 10.10.10.132 = softphone1 registered from public IP
> 
> 11.11.11.= softphone2 registered from public IP
> 
> 10.10.10.138 = FS

Those are private addresses; there's a NAT device involved somewhere, and
that's likely to be where your problems are.



From infos at madovsky.org  Sun Mar 21 22:55:17 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 01:55:17 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <896696322ADB4DDEA77C06E3C5D5B676@MOBILEE1705><A1805ACA-F025-458E-9F36-E1A76E48A160@freeswitch.org><98FD50583C84472EB52EAF12CCA015C9@MOBILEE1705><F936721E-D82A-48C9-A185-532298CD5C03@freeswitch.org><DB233188E7704AC5AC41F1F64DF165BC@MOBILEE1705><428CE3BB-06FE-4C46-AB14-CD9E6E961B26@freeswitch.org><79A015FDD1C443B4B18CFA15941916DF@MOBILEE1705>
	<20100322054920.GB10155@jdc.jasonjgw.net>
Message-ID: <CFD5777F32D34733991F6C0B099C9EF5@MOBILEE1705>

I replace public ip by these fake one...

----- Original Message ----- 
From: "Jason White" <jason at jasonjgw.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 22, 2010 1:49 AM
Subject: Re: [Freeswitch-users] rtp timeout yet


> Madovsky <infos at madovsky.org> wrote:
>> result with 
>> 
>> tcpdump -ennqti br0 -T rtp -vvv dst port 5060
>> 
>> 10.10.10.132 = softphone1 registered from public IP
>> 
>> 11.11.11.= softphone2 registered from public IP
>> 
>> 10.10.10.138 = FS
> 
> Those are private addresses; there's a NAT device involved somewhere, and
> that's likely to be where your problems are.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Sun Mar 21 23:31:39 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 02:31:39 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <B720C7D1F11C468AAA3720B96EF35326@MOBILEE1705>

I deactivated iptables, no router, no nat,
only 2 public IP, 2 softphones connected on the same FS which is on a public
IP to.
default config, result -> rtp timeout
crazy that's not working since I can make 2 softphones behind a firewall
work....

ok that's enough for me tonight ! ;)

----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 22, 2010 1:55 AM
Subject: Re: [Freeswitch-users] rtp timeout yet


>I replace public ip by these fake one...
>
> ----- Original Message ----- 
> From: "Jason White" <jason at jasonjgw.net>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, March 22, 2010 1:49 AM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
>> Madovsky <infos at madovsky.org> wrote:
>>> result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060
>>>
>>> 10.10.10.132 = softphone1 registered from public IP
>>>
>>> 11.11.11.= softphone2 registered from public IP
>>>
>>> 10.10.10.138 = FS
>>
>> Those are private addresses; there's a NAT device involved somewhere, and
>> that's likely to be where your problems are.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 



From kees at mroffice.org  Mon Mar 22 00:43:49 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Mon, 22 Mar 2010 20:43:49 +1300
Subject: [Freeswitch-users] Compiling freeswitch on Ubuntu 10.04
Message-ID: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>

Hi there,

I am trying to compile Freeswitch on Ubuntu 10.04 but am getting stranded as
follows:

g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crti.o: No such file
or directory
g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtbeginS.o: No such file or
directory
g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtendS.o: No such file or directory
g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crtn.o: No such file
or directory
make[2]: *** [libfreeswitch.la] Error 1
make[2]: Leaving directory `/usr/src/freeswitch.trunk'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch.trunk'
make: *** [freeswitch] Error 2


I think this is related to the gcc version; ubuntu 10.04 comes with version
4.4.3 rather than 4.4.1.

Does anyone know how to fix this?
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From brian at microcomaustralia.com.au  Mon Mar 22 01:15:24 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Mar 2010 19:15:24 +1100
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <5800526b1003211939g1a9cf076m13638514d4ebfde9@mail.gmail.com>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
	<5800526b1003211939g1a9cf076m13638514d4ebfde9@mail.gmail.com>
Message-ID: <3c5cf5261003220115mec6b457s89601636d492270b@mail.gmail.com>

On 22 March 2010 13:39, Carlos Talbot <carlos.talbot at gmail.com> wrote:
> I've experienced the same problem. I have several SIP clients registered to
> one extension on my FreeSWITCH server at home (multiple-registrations). If
> one of the clients happens to be siproid it will immediately answer while
> the other clients stop ringing. It also sends a European ringback tone to
> the caller for some reason. I'm still trying to debug the sipdroid code to
> figure out where this is happening.

Looks like sipdroid might be in the clear.

107.338991 202.173.153.98 -> 192.168.200.42 SIP/SDP Request: INVITE
sip:1002 at 192.168.200.42:43658;transport=udp, with session description
107.554359 192.168.200.42 -> 202.173.153.98 SIP Status: 100 Trying
108.387799 192.168.200.42 -> 202.173.153.98 SIP/SDP Status: 180
Ringing, with session description

<hang up>

121.372867 202.173.153.98 -> 192.168.200.42 SIP Request: CANCEL
sip:1002 at 192.168.200.42:43658;transport=udp
121.496434 192.168.200.42 -> 202.173.153.98 SIP Status: 200 OK
121.559057 192.168.200.42 -> 202.173.153.98 SIP Status: 487 Request Terminated
121.560015 202.173.153.98 -> 192.168.200.42 SIP Request: ACK
sip:1002 at 192.168.200.42:43658;transport=udp
123.679219 192.168.200.42 -> 202.173.153.98 UDP Source port: 43658
Destination port: sip

This gets me back to the other thread - "How long did the phone ring?"
which discusses 180 vs 183 styles of answering calls. In this case it
is using 180 which was described as "your phone rings (Ringing
message)", not 183 as I expected it might be which is "early media or
inband ringing (Session Progress message)". All of the above looks
perfect to me.

However something is really screwy. First question, what generates
that non-standard ring tone I can hear?

At this time freeswitch shows:

freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
994c5126-d961-4838-b642-d41060242283,inbound,2010-03-22
18:52:22,1269244342,sofia/internal/1000 at 202.173.153.98,CS_EXECUTE,Lounge,1000,192.168.87.13,1002,bridge,user/1002 at microcomaustralia.com.au,XML,default,PCMA,8000,PCMA,8000,,huey,1000 at 202.173.153.98,
bef34f96-bb3c-44b2-b8d8-41f5a1e7913e,outbound,2010-03-22
18:52:22,1269244342,sofia/internal/sip:1002 at 192.168.200.42:43658,CS_EXCHANGE_MEDIA,Lounge,1001,192.168.87.13,1002,,,XML,default,PCMA,8000,PCMA,8000,,huey,1002 at microcomaustralia.com.au,

which seems to imply that it is passing audio data to/from siphoid.
However nothing shows up on the sip port for negotiating this. How is
this possible? It seems like I am getting inband ringing with 180. Am
I confused?

I also note RTP packets whizzing past, which confirms this.

I have added full debugging from freeswitch to the end. I haven't yet
analysed it in detail.

> My impression is that the primary developer of sipdroid is in an Asterisk
> world and so does not really care on how the mjsip based SIP stack behaves
> in other environments.

:-(

> Incidentally, Brian West had replied to your email regarding a copy of
> sipdroid I recently built with the SILK codec. I've recently checked in the
> code along with BV support. I recommend SILK over BV with sipdroid though as
> BV is a CPU hog. Well, this is partly to do with how sipdroid behaves too.
> See here:?http://code.google.com/p/sipdroid/issues/detail?id=358
> The patch includes SILK's?wide band codecs so it's worth testing against
> a?FreeSWITCH server. I'd like others feedback on the perceived sound quality
> (especially those with Droids as I hear they have a pretty decent speaker).
> On my HTC magic you can tell the difference between SILK8 and 16 but I'm not
> that impressed on the fidelity of the?handset speaker. Most cell phone
> manufactures nowadays (including Apple) don't seem to care about the primary
> purpose of the device; a decent sounding phone call.

Ok, will look at codecs once I work out what is going on with ringing...



2010-03-22 18:52:22.120480 [DEBUG] sofia.c:5797 IP 192.168.87.13
Approved by acl "domains[1001 at microcomaustralia.com.au]". Access
Granted.
2010-03-22 18:52:22.120480 [DEBUG] sofia.c:5919 Authenticating user
1001 at microcomaustralia.com.au
2010-03-22 18:52:22.120480 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/1000 at 202.173.153.98
[994c5126-d961-4838-b642-d41060242283]
2010-03-22 18:52:22.151167 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_NEW
2010-03-22 18:52:22.151167 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1000 at 202.173.153.98) State NEW
2010-03-22 18:52:22.171563 [DEBUG] sofia.c:4152 Channel
sofia/internal/1000 at 202.173.153.98 entering state [received][100]
2010-03-22 18:52:22.171563 [DEBUG] sofia.c:4163 Remote SDP:
v=0
o=- 3092896 3092896 IN IP4 192.168.87.13
s=-
c=IN IP4 192.168.87.13
t=0 0
m=audio 16468 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30

2010-03-22 18:52:22.171563 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:30]/[G7221:115:32000:20]
2010-03-22 18:52:22.171563 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:30]/[G7221:107:16000:20]
2010-03-22 18:52:22.178570 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:30]/[PCMU:0:8000:20]
2010-03-22 18:52:22.178570 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:30]/[PCMA:8:8000:20]
2010-03-22 18:52:22.178570 [DEBUG] sofia_glue.c:3626 Substituting
codec PCMA at 30i@8000h
2010-03-22 18:52:22.181212 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1000 at 202.173.153.98 PCMA/8000 30 ms 240 samples
2010-03-22 18:52:22.187060 [DEBUG] sofia_glue.c:3518 Set 2833 dtmf
send payload to 101
2010-03-22 18:52:22.187060 [DEBUG] sofia.c:4309
(sofia/internal/1000 at 202.173.153.98) State Change CS_NEW -> CS_INIT
2010-03-22 18:52:22.187060 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:22.187060 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_INIT
2010-03-22 18:52:22.187060 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 202.173.153.98) State INIT
2010-03-22 18:52:22.187060 [DEBUG] mod_sofia.c:83
sofia/internal/1000 at 202.173.153.98 SOFIA INIT
2010-03-22 18:52:22.187060 [DEBUG] mod_sofia.c:117
(sofia/internal/1000 at 202.173.153.98) State Change CS_INIT ->
CS_ROUTING
2010-03-22 18:52:22.187060 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:22.187060 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 202.173.153.98) State INIT going to sleep
2010-03-22 18:52:22.187060 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_ROUTING
2010-03-22 18:52:22.198995 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000 at 202.173.153.98) State ROUTING
2010-03-22 18:52:22.198995 [DEBUG] mod_sofia.c:140
sofia/internal/1000 at 202.173.153.98 SOFIA ROUTING
2010-03-22 18:52:22.198995 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000 at 202.173.153.98 Standard ROUTING
2010-03-22 18:52:22.198995 [INFO] mod_dialplan_xml.c:418 Processing
Lounge->1002 in context default
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->global-intercept] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[global-intercept] destination_number(1002) =~ /^886$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[group-intercept] destination_number(1002) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[intercept-ext] destination_number(1002) =~ /^\*\*(\d+)$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing [default->redial]
continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [redial]
destination_number(1002) =~ /^870$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing [default->global]
continue=true
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [global]
${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/1000 at 202.173.153.98 Absolute Condition [global]
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->snom-demo-2] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[snom-demo-2] destination_number(1002) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->snom-demo-1] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[snom-demo-1] destination_number(1002) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [eavesdrop]
destination_number(1002) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->eavesdrop] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [eavesdrop]
destination_number(1002) =~ /^779$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->call_return] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[call_return] destination_number(1002) =~ /^\*69$|^869$|^lcr$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->del-group] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [del-group]
destination_number(1002) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->add-group] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL) [add-group]
destination_number(1002) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[call-group-simo] destination_number(1002) =~ /^82(\d{2})$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[call-group-order] destination_number(1002) =~ /^83(\d{2})$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->extension-intercom] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (FAIL)
[extension-intercom] destination_number(1002) =~ /^8(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/1000 at 202.173.153.98 Regex (PASS)
[Local_Extension] destination_number(1002) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1000 at 202.173.153.98 Action set(dialed_extension=1002)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
export(dialed_extension=1002)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action bind_meta_app(1 b
s execute_extension::dx XML features)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action bind_meta_app(2 b
s record_session::/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action bind_meta_app(3 b
s execute_extension::cf XML features)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action set(ringback=${au-ring})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
set(transfer_ringback=local_stream://moh)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action set(call_timeout=30)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action
bridge(user/${dialed_extension}@${domain_name})
Dialplan: sofia/internal/1000 at 202.173.153.98 Action answer()
Dialplan: sofia/internal/1000 at 202.173.153.98 Action sleep(1000)
Dialplan: sofia/internal/1000 at 202.173.153.98 Action voicemail(default
${domain_name} ${dialed_extension})
2010-03-22 18:52:22.248545 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/1000 at 202.173.153.98) State Change CS_ROUTING ->
CS_EXECUTE
2010-03-22 18:52:22.248545 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:22.248545 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000 at 202.173.153.98) State ROUTING going to sleep
2010-03-22 18:52:22.250461 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_EXECUTE
2010-03-22 18:52:22.254446 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/1000 at 202.173.153.98) State EXECUTE
2010-03-22 18:52:22.254446 [DEBUG] mod_sofia.c:226
sofia/internal/1000 at 202.173.153.98 SOFIA EXECUTE
2010-03-22 18:52:22.254446 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 202.173.153.98 Standard EXECUTE
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-spymap/1000/994c5126-d961-4838-b642-d41060242283)
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-last_dial/1000/1002)
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-last_dial/global/994c5126-d961-4838-b642-d41060242283)
EXECUTE sofia/internal/1000 at 202.173.153.98 set(dialed_extension=1002)
2010-03-22 18:52:22.277534 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET [dialed_extension]=[1002]
EXECUTE sofia/internal/1000 at 202.173.153.98 export(dialed_extension=1002)
2010-03-22 18:52:22.311503 [DEBUG] mod_dptools.c:898 EXPORT
[dialed_extension]=[1002]
EXECUTE sofia/internal/1000 at 202.173.153.98 bind_meta_app(1 b s
execute_extension::dx XML features)
2010-03-22 18:52:22.323369 [INFO] switch_ivr_async.c:2339 Bound B-Leg:
1 execute_extension::dx XML features
EXECUTE sofia/internal/1000 at 202.173.153.98 bind_meta_app(2 b s
record_session::/opt/freeswitch/recordings/1000.2010-03-22-18-52-22.wav)
2010-03-22 18:52:22.337501 [INFO] switch_ivr_async.c:2339 Bound B-Leg:
2 record_session::/opt/freeswitch/recordings/1000.2010-03-22-18-52-22.wav
EXECUTE sofia/internal/1000 at 202.173.153.98 bind_meta_app(3 b s
execute_extension::cf XML features)
2010-03-22 18:52:22.354499 [INFO] switch_ivr_async.c:2339 Bound B-Leg:
3 execute_extension::cf XML features
EXECUTE sofia/internal/1000 at 202.173.153.98 set(ringback=)
2010-03-22 18:52:22.359488 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET [ringback]=[UNDEF]
EXECUTE sofia/internal/1000 at 202.173.153.98
set(transfer_ringback=local_stream://moh)
2010-03-22 18:52:22.372662 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET
[transfer_ringback]=[local_stream://moh]
EXECUTE sofia/internal/1000 at 202.173.153.98 set(call_timeout=30)
2010-03-22 18:52:22.384505 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET [call_timeout]=[30]
EXECUTE sofia/internal/1000 at 202.173.153.98 set(hangup_after_bridge=true)
2010-03-22 18:52:22.398395 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1000 at 202.173.153.98 set(continue_on_fail=true)
2010-03-22 18:52:22.403370 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-call_return/1002/1000)
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-last_dial_ext/1002/994c5126-d961-4838-b642-d41060242283)
EXECUTE sofia/internal/1000 at 202.173.153.98 set(called_party_callgroup=)
2010-03-22 18:52:22.437699 [DEBUG] mod_dptools.c:814
sofia/internal/1000 at 202.173.153.98 SET
[called_party_callgroup]=[UNDEF]
EXECUTE sofia/internal/1000 at 202.173.153.98
hash(insert/microcomaustralia.com.au-last_dial//994c5126-d961-4838-b642-d41060242283)
EXECUTE sofia/internal/1000 at 202.173.153.98
bridge(user/1002 at microcomaustralia.com.au)
2010-03-22 18:52:22.505426 [DEBUG] switch_ivr_originate.c:1885
variable string 0 = [presence_id=1002 at microcomaustralia.com.au]
2010-03-22 18:52:22.505426 [NOTICE] switch_channel.c:669 New Channel
sofia/internal/sip:1002 at 192.168.200.42:43658
[bef34f96-bb3c-44b2-b8d8-41f5a1e7913e]
2010-03-22 18:52:22.509487 [DEBUG] mod_sofia.c:3375
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change CS_NEW ->
CS_INIT
2010-03-22 18:52:22.509487 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:22.521870 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_INIT
2010-03-22 18:52:22.521870 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1002 at 192.168.200.42:43658) State INIT
2010-03-22 18:52:22.521870 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1002 at 192.168.200.42:43658 SOFIA INIT
2010-03-22 18:52:22.532449 [DEBUG] mod_sofia.c:117
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change CS_INIT ->
CS_ROUTING
2010-03-22 18:52:22.532449 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:22.532449 [DEBUG] sofia.c:4152 Channel
sofia/internal/sip:1002 at 192.168.200.42:43658 entering state
[calling][0]
2010-03-22 18:52:22.532449 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1002 at 192.168.200.42:43658) State INIT going to
sleep
2010-03-22 18:52:22.536471 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_ROUTING
2010-03-22 18:52:22.542405 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1002 at 192.168.200.42:43658) State ROUTING
2010-03-22 18:52:22.542405 [DEBUG] mod_sofia.c:140
sofia/internal/sip:1002 at 192.168.200.42:43658 SOFIA ROUTING
2010-03-22 18:52:22.542405 [DEBUG] switch_ivr_originate.c:66
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change CS_ROUTING
-> CS_CONSUME_MEDIA
2010-03-22 18:52:22.542405 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:22.545489 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/sip:1002 at 192.168.200.42:43658) State ROUTING going to
sleep
2010-03-22 18:52:22.545489 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_CONSUME_MEDIA
2010-03-22 18:52:22.549537 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1002 at 192.168.200.42:43658) State CONSUME_MEDIA
2010-03-22 18:52:22.549537 [DEBUG] switch_core_state_machine.c:360
(sofia/internal/sip:1002 at 192.168.200.42:43658) State CONSUME_MEDIA
going to sleep
2010-03-22 18:52:22.549537 [INFO] ozmod_zt.c:638 Setting echo cancel
to 64 taps for 1:1
2010-03-22 18:52:22.553504 [WARNING] ozmod_zt.c:644 Echo cancel not
available for 1:1
2010-03-22 18:52:22.553504 [DEBUG] mod_openzap.c:403 Set codec PCMU 20ms
2010-03-22 18:52:22.559427 [DEBUG] mod_openzap.c:1312 Connect outbound
channel OpenZAP/1:1/
2010-03-22 18:52:22.562442 [NOTICE] switch_channel.c:669 New Channel
OpenZAP/1:1/ [63bd7875-06a6-4779-9c21-f69d08a1e6f7]
2010-03-22 18:52:22.562442 [DEBUG] mod_openzap.c:1326 (OpenZAP/1:1/)
State Change CS_NEW -> CS_INIT
2010-03-22 18:52:22.562442 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:22.562442 [DEBUG] ozmod_analog.c:80 Changing state on
1:1 from DOWN to GENRING
2010-03-22 18:52:22.562442 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL
thread starting.
2010-03-22 18:52:22.566556 [INFO] ozmod_zt.c:638 Setting echo cancel
to 64 taps for 1:1
2010-03-22 18:52:22.566556 [WARNING] ozmod_zt.c:644 Echo cancel not
available for 1:1
2010-03-22 18:52:22.566556 [DEBUG] ozmod_analog.c:450 Executing state
handler on 1:1 for GENRING
2010-03-22 18:52:22.571354 [DEBUG] mod_openzap.c:1593 got FXS sig [PROGRESS]
2010-03-22 18:52:22.571354 [NOTICE] mod_openzap.c:1609 Ring-Ready OpenZAP/1:1/!
2010-03-22 18:52:22.583992 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/1:1/) Running State Change CS_INIT
2010-03-22 18:52:22.583992 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/1:1/) State INIT
2010-03-22 18:52:22.587618 [DEBUG] mod_openzap.c:431 (OpenZAP/1:1/)
State Change CS_INIT -> CS_ROUTING
2010-03-22 18:52:22.587618 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:22.587618 [DEBUG] switch_core_state_machine.c:338
(OpenZAP/1:1/) State INIT going to sleep
2010-03-22 18:52:22.589860 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/1:1/) Running State Change CS_ROUTING
2010-03-22 18:52:22.599032 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/1:1/) State ROUTING
2010-03-22 18:52:22.599032 [DEBUG] mod_openzap.c:454 OpenZAP/1:1/
CHANNEL ROUTING
2010-03-22 18:52:22.599032 [DEBUG] switch_ivr_originate.c:66
(OpenZAP/1:1/) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-03-22 18:52:22.599032 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:22.602467 [DEBUG] switch_core_state_machine.c:341
(OpenZAP/1:1/) State ROUTING going to sleep
2010-03-22 18:52:22.602467 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/1:1/) Running State Change CS_CONSUME_MEDIA
2010-03-22 18:52:22.606517 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/1:1/) State CONSUME_MEDIA
2010-03-22 18:52:22.606517 [DEBUG] switch_core_state_machine.c:360
(OpenZAP/1:1/) State CONSUME_MEDIA going to sleep
2010-03-22 18:52:22.611986 [NOTICE] mod_sofia.c:1832 Ring-Ready
sofia/internal/1000 at 202.173.153.98!
2010-03-22 18:52:22.618469 [DEBUG] sofia.c:4152 Channel
sofia/internal/1000 at 202.173.153.98 entering state [early][180]
2010-03-22 18:52:22.618469 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:22.618469 [NOTICE] switch_ivr_originate.c:437 Ring
Ready sofia/internal/1000 at 202.173.153.98!
2010-03-22 18:52:23.599391 [INFO] sofia.c:662 Update Callee ID to "1002" <1002>
2010-03-22 18:52:23.616441 [DEBUG] sofia.c:4152 Channel
sofia/internal/sip:1002 at 192.168.200.42:43658 entering state
[proceeding][180]
2010-03-22 18:52:23.616441 [DEBUG] sofia.c:4163 Remote SDP:
v=0
o=1002 at microcomaustralia.com.au 0 0 IN IP4 192.168.200.42
s=Session SIP/SDP
c=IN IP4 192.168.200.42
t=0 0
m=audio 21000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:30]
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:20]/[PCMU:0:8000:30]
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:3579 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:30]
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:3626 Substituting
codec PCMA at 20i@8000h
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/sip:1002 at 192.168.200.42:43658 PCMA/8000 20 ms 160
samples
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:3512 Set 2833 dtmf
recv payload to 0
2010-03-22 18:52:23.616441 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/internal/sip:1002 at 192.168.200.42:43658] 202.173.153.98 port
26012 -> 192.168.200.42 port 21000 codec: 8 ms: 20
2010-03-22 18:52:23.616441 [DEBUG] switch_rtp.c:1182 Starting timer
[soft] 160 bytes per 20ms
2010-03-22 18:52:23.625834 [NOTICE] sofia_glue.c:3150 Pre-Answer
sofia/internal/sip:1002 at 192.168.200.42:43658!
2010-03-22 18:52:23.625834 [DEBUG] switch_channel.c:2180 Send signal
sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:23.625834 [NOTICE] switch_ivr_originate.c:3079 Hangup
OpenZAP/1:1/ [CS_CONSUME_MEDIA] [LOSE_RACE]
2010-03-22 18:52:23.630894 [DEBUG] switch_channel.c:2071 Send signal
OpenZAP/1:1/ [KILL]
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/1:1/) Running State Change CS_HANGUP
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:499
(OpenZAP/1:1/) State HANGUP
2010-03-22 18:52:23.630894 [DEBUG] mod_openzap.c:526 Changing state on
1:1 from GENRING to HANGUP
2010-03-22 18:52:23.630894 [DEBUG] mod_openzap.c:556 OpenZAP/1:1/ CHANNEL HANGUP
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:46
OpenZAP/1:1/ Standard HANGUP, cause: LOSE_RACE
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:499
(OpenZAP/1:1/) State HANGUP going to sleep
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:333
(OpenZAP/1:1/) State Change CS_HANGUP -> CS_REPORTING
2010-03-22 18:52:23.630894 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:314
(OpenZAP/1:1/) Running State Change CS_REPORTING
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:590
(OpenZAP/1:1/) State REPORTING
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:53
OpenZAP/1:1/ Standard REPORTING, cause: LOSE_RACE
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:590
(OpenZAP/1:1/) State REPORTING going to sleep
2010-03-22 18:52:23.630894 [DEBUG] switch_core_state_machine.c:327
(OpenZAP/1:1/) State Change CS_REPORTING -> CS_DESTROY
2010-03-22 18:52:23.630894 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:23.630894 [DEBUG] switch_core_session.c:1161 Session
84 (OpenZAP/1:1/) Locked, Waiting on external entities
2010-03-22 18:52:23.640155 [DEBUG] switch_core_session.c:1018 Send
signal OpenZAP/1:1/ [BREAK]
2010-03-22 18:52:23.646445 [DEBUG] ozmod_analog.c:450 Executing state
handler on 1:1 for HANGUP
2010-03-22 18:52:23.651492 [INFO] switch_ivr_originate.c:3105 Sending
early media
2010-03-22 18:52:23.653336 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/internal/1000 at 202.173.153.98] 202.173.153.98 port 23450 ->
192.168.87.13 port 16468 codec: 8 ms: 30
2010-03-22 18:52:23.654347 [DEBUG] switch_rtp.c:1182 Starting timer
[soft] 240 bytes per 30ms
2010-03-22 18:52:23.657559 [DEBUG] mod_sofia.c:1901 Ring SDP:
v=0
o=FreeSWITCH 1269220893 1269220894 IN IP4 202.173.153.98
s=FreeSWITCH
c=IN IP4 202.173.153.98
t=0 0
m=audio 23450 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv

2010-03-22 18:52:23.659401 [NOTICE] mod_sofia.c:1904 Pre-Answer
sofia/internal/1000 at 202.173.153.98!
2010-03-22 18:52:23.668549 [DEBUG] sofia.c:4147 Channel
sofia/internal/1000 at 202.173.153.98 skipping state [early][183]
2010-03-22 18:52:23.684459 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:23.684459 [DEBUG] switch_ivr_originate.c:3148
Originate Resulted in Success:
[sofia/internal/sip:1002 at 192.168.200.42:43658]
2010-03-22 18:52:23.684459 [NOTICE] switch_core_session.c:1179 Session
84 (OpenZAP/1:1/) Ended
2010-03-22 18:52:23.684459 [NOTICE] switch_core_session.c:1181 Close
Channel OpenZAP/1:1/ [CS_DESTROY]
2010-03-22 18:52:23.684459 [DEBUG] switch_core_state_machine.c:428
(OpenZAP/1:1/) Running State Change CS_DESTROY
2010-03-22 18:52:23.684459 [DEBUG] switch_core_state_machine.c:439
(OpenZAP/1:1/) State DESTROY
2010-03-22 18:52:23.684459 [DEBUG] switch_core_state_machine.c:60
OpenZAP/1:1/ Standard DESTROY
2010-03-22 18:52:23.684459 [DEBUG] switch_core_state_machine.c:439
(OpenZAP/1:1/) State DESTROY going to sleep
2010-03-22 18:52:23.699411 [DEBUG] switch_ivr_originate.c:3148
Originate Resulted in Success:
[sofia/internal/sip:1002 at 192.168.200.42:43658]
2010-03-22 18:52:23.705505 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:23.705505 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:23.705505 [DEBUG] switch_ivr_bridge.c:1182
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2010-03-22 18:52:23.709595 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:23.709595 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_EXCHANGE_MEDIA
2010-03-22 18:52:23.709595 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:1002 at 192.168.200.42:43658) State EXCHANGE_MEDIA
2010-03-22 18:52:23.709595 [DEBUG] mod_sofia.c:504 SOFIA EXCHANGE_MEDIA
2010-03-22 18:52:23.776426 [DEBUG] switch_rtp.c:2064 Correct ip/port confirmed.
2010-03-22 18:52:23.776426 [DEBUG] switch_core_io.c:896 Engaging Write
Buffer at 320 bytes to accommodate 480->320
2010-03-22 18:52:24.165340 [DEBUG] ozmod_analog.c:391 Changing state
on 1:1 from HANGUP to DOWN
2010-03-22 18:52:24.185341 [DEBUG] ozmod_analog.c:450 Executing state
handler on 1:1 for DOWN
2010-03-22 18:52:24.185341 [DEBUG] mod_openzap.c:1593 got FXS sig [STOP]
2010-03-22 18:52:24.185341 [DEBUG] zap_io.c:1383 channel done 1:1
2010-03-22 18:52:24.185341 [DEBUG] ozmod_analog.c:764 ANALOG CHANNEL
1:1 thread ended.
2010-03-22 18:52:24.295343 [DEBUG] switch_rtp.c:2064 Correct ip/port confirmed.
2010-03-22 18:52:24.295343 [DEBUG] switch_core_io.c:896 Engaging Write
Buffer at 480 bytes to accommodate 320->480

freeswitch at internal>
freeswitch at internal> show cha

[            channels]	[                chat]	


freeswitch at internal> show channels
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure,hostname,presence_id,presence_data
994c5126-d961-4838-b642-d41060242283,inbound,2010-03-22
18:52:22,1269244342,sofia/internal/1000 at 202.173.153.98,CS_EXECUTE,Lounge,1000,192.168.87.13,1002,bridge,user/1002 at microcomaustralia.com.au,XML,default,PCMA,8000,PCMA,8000,,huey,1000 at 202.173.153.98,
bef34f96-bb3c-44b2-b8d8-41f5a1e7913e,outbound,2010-03-22
18:52:22,1269244342,sofia/internal/sip:1002 at 192.168.200.42:43658,CS_EXCHANGE_MEDIA,Lounge,1001,192.168.87.13,1002,,,XML,default,PCMA,8000,PCMA,8000,,huey,1002 at microcomaustralia.com.au,

2 total.

freeswitch at internal> 2010-03-22 18:52:39.786251 [DEBUG] sofia.c:4152
Channel sofia/internal/1000 at 202.173.153.98 entering state
[terminated][487]
2010-03-22 18:52:39.786251 [NOTICE] sofia.c:4788 Hangup
sofia/internal/1000 at 202.173.153.98 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2010-03-22 18:52:39.786251 [DEBUG] switch_channel.c:2071 Send signal
sofia/internal/1000 at 202.173.153.98 [KILL]
2010-03-22 18:52:39.786251 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:39.795756 [DEBUG] switch_ivr_bridge.c:472
sofia/internal/1000 at 202.173.153.98 ending bridge by request from write
function
2010-03-22 18:52:39.795756 [DEBUG] switch_core_session.c:638 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:39.795756 [DEBUG] switch_ivr_bridge.c:565 BRIDGE
THREAD DONE [sofia/internal/sip:1002 at 192.168.200.42:43658]
2010-03-22 18:52:39.795756 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:39.795756 [NOTICE] switch_ivr_bridge.c:637 Hangup
sofia/internal/sip:1002 at 192.168.200.42:43658 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2010-03-22 18:52:39.795756 [DEBUG] switch_channel.c:2071 Send signal
sofia/internal/sip:1002 at 192.168.200.42:43658 [KILL]
2010-03-22 18:52:39.795756 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:39.795756 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:1002 at 192.168.200.42:43658) State EXCHANGE_MEDIA
going to sleep
2010-03-22 18:52:39.795756 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_HANGUP
2010-03-22 18:52:39.795756 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/sip:1002 at 192.168.200.42:43658) State HANGUP
2010-03-22 18:52:39.795756 [DEBUG] mod_sofia.c:405
sofia/internal/sip:1002 at 192.168.200.42:43658 Overriding SIP cause 480
with 487 from the other leg
2010-03-22 18:52:39.795756 [DEBUG] mod_sofia.c:411 Channel
sofia/internal/sip:1002 at 192.168.200.42:43658 hanging up, cause:
NORMAL_CLEARING
2010-03-22 18:52:39.795756 [DEBUG] switch_ivr_bridge.c:478
sofia/internal/1000 at 202.173.153.98 ending bridge by request from read
function
2010-03-22 18:52:39.852735 [DEBUG] switch_ivr_bridge.c:565 BRIDGE
THREAD DONE [sofia/internal/1000 at 202.173.153.98]
2010-03-22 18:52:39.852735 [DEBUG] switch_ivr_bridge.c:585 Send signal
sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:39.852735 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/1000 at 202.173.153.98) State EXECUTE going to sleep
2010-03-22 18:52:39.852735 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_HANGUP
2010-03-22 18:52:39.852735 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/1000 at 202.173.153.98) State HANGUP
2010-03-22 18:52:39.852735 [DEBUG] mod_sofia.c:405
sofia/internal/1000 at 202.173.153.98 Overriding SIP cause 487 with 487
from the other leg
2010-03-22 18:52:39.852735 [DEBUG] mod_sofia.c:411 Channel
sofia/internal/1000 at 202.173.153.98 hanging up, cause:
ORIGINATOR_CANCEL
2010-03-22 18:52:39.902442 [DEBUG] mod_sofia.c:464 Sending CANCEL to
sofia/internal/sip:1002 at 192.168.200.42:43658
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1000 at 202.173.153.98 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/1000 at 202.173.153.98) State HANGUP going to sleep
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/1000 at 202.173.153.98) State Change CS_HANGUP ->
CS_REPORTING
2010-03-22 18:52:39.910280 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_REPORTING
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/1000 at 202.173.153.98) State REPORTING
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:53
sofia/internal/1000 at 202.173.153.98 Standard REPORTING, cause:
ORIGINATOR_CANCEL
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/1000 at 202.173.153.98) State REPORTING going to sleep
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/1000 at 202.173.153.98) State Change CS_REPORTING ->
CS_DESTROY
2010-03-22 18:52:39.910280 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/1000 at 202.173.153.98 [BREAK]
2010-03-22 18:52:39.910280 [DEBUG] switch_core_session.c:1161 Session
82 (sofia/internal/1000 at 202.173.153.98) Locked, Waiting on external
entities
2010-03-22 18:52:39.910280 [NOTICE] switch_core_session.c:1179 Session
82 (sofia/internal/1000 at 202.173.153.98) Ended
2010-03-22 18:52:39.910280 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/internal/1000 at 202.173.153.98 [CS_DESTROY]
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:428
(sofia/internal/1000 at 202.173.153.98) Running State Change CS_DESTROY
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/1000 at 202.173.153.98) State DESTROY
2010-03-22 18:52:39.910280 [DEBUG] mod_sofia.c:338
sofia/internal/1000 at 202.173.153.98 SOFIA DESTROY
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1000 at 202.173.153.98 Standard DESTROY
2010-03-22 18:52:39.910280 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/1000 at 202.173.153.98) State DESTROY going to sleep
2010-03-22 18:52:39.925570 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1002 at 192.168.200.42:43658 Standard HANGUP, cause:
NORMAL_CLEARING
2010-03-22 18:52:39.925570 [DEBUG] switch_core_state_machine.c:499
(sofia/internal/sip:1002 at 192.168.200.42:43658) State HANGUP going to
sleep
2010-03-22 18:52:39.931493 [DEBUG] switch_core_state_machine.c:333
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change CS_HANGUP
-> CS_REPORTING
2010-03-22 18:52:39.931493 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:39.931493 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_REPORTING
2010-03-22 18:52:39.931493 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/sip:1002 at 192.168.200.42:43658) State REPORTING
2010-03-22 18:52:39.931493 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:1002 at 192.168.200.42:43658 Standard REPORTING,
cause: NORMAL_CLEARING
2010-03-22 18:52:39.935435 [DEBUG] switch_core_state_machine.c:590
(sofia/internal/sip:1002 at 192.168.200.42:43658) State REPORTING going
to sleep
2010-03-22 18:52:39.935435 [DEBUG] switch_core_state_machine.c:327
(sofia/internal/sip:1002 at 192.168.200.42:43658) State Change
CS_REPORTING -> CS_DESTROY
2010-03-22 18:52:39.935435 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/sip:1002 at 192.168.200.42:43658 [BREAK]
2010-03-22 18:52:39.937446 [DEBUG] switch_core_session.c:1161 Session
83 (sofia/internal/sip:1002 at 192.168.200.42:43658) Locked, Waiting on
external entities
2010-03-22 18:52:39.937446 [NOTICE] switch_core_session.c:1179 Session
83 (sofia/internal/sip:1002 at 192.168.200.42:43658) Ended
2010-03-22 18:52:39.937446 [NOTICE] switch_core_session.c:1181 Close
Channel sofia/internal/sip:1002 at 192.168.200.42:43658 [CS_DESTROY]
2010-03-22 18:52:39.941335 [DEBUG] switch_core_state_machine.c:428
(sofia/internal/sip:1002 at 192.168.200.42:43658) Running State Change
CS_DESTROY
2010-03-22 18:52:39.941335 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/sip:1002 at 192.168.200.42:43658) State DESTROY
2010-03-22 18:52:39.941335 [DEBUG] mod_sofia.c:338
sofia/internal/sip:1002 at 192.168.200.42:43658 SOFIA DESTROY
2010-03-22 18:52:39.941335 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:1002 at 192.168.200.42:43658 Standard DESTROY
2010-03-22 18:52:39.941335 [DEBUG] switch_core_state_machine.c:439
(sofia/internal/sip:1002 at 192.168.200.42:43658) State DESTROY going to
sleep



-- 
Brian May <brian at microcomaustralia.com.au>


From brian at microcomaustralia.com.au  Mon Mar 22 01:24:59 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Mar 2010 19:24:59 +1100
Subject: [Freeswitch-users] auth incoming calls
Message-ID: <3c5cf5261003220124g20f065au4f60196bdbbe82c0@mail.gmail.com>

Hello,

In debugging that previous problem I noticed another problem.

How do I get freeswitch to authenticate incomming calls? As far as I
can tell it is not working:

483.970655 192.168.87.13 -> 202.173.153.98 SIP/SDP Request: INVITE
sip:9999 at 202.173.153.98, with session description
483.972989 202.173.153.98 -> 192.168.87.13 SIP Status: 100 Trying
484.180018 202.173.153.98 -> 192.168.87.13 SIP/SDP Status: 200 OK,
with session description
484.208932 192.168.87.13 -> 202.173.153.98 SIP Request: ACK
sip:9999 at 202.173.153.98:5060;transport=udp
490.634429 192.168.87.13 -> 202.173.153.98 SIP Request: BYE
sip:9999 at 202.173.153.98:5060;transport=udp
490.637862 202.173.153.98 -> 192.168.87.13 SIP Status: 200 OK

Yet the setting is appears to be enabled:

<param name="auth-calls" value="true"/>

-- 
Brian May <brian at microcomaustralia.com.au>


From gmaruzz at celliax.org  Mon Mar 22 01:25:59 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Mar 2010 09:25:59 +0100
Subject: [Freeswitch-users] Compiling freeswitch on Ubuntu 10.04
In-Reply-To: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>
References: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>
Message-ID: <7b197bef1003220125x408cc328qfd952addb5280a0b@mail.gmail.com>

On Mon, Mar 22, 2010 at 8:43 AM, Kees Varekamp <kees at mroffice.org> wrote:
> Hi there,
> I am trying to compile Freeswitch on Ubuntu 10.04 but am getting stranded as

Kees, you're trying to use a non released (alpha, whatever) Ubuntu
version to build what is intended to be a solid server for telephony.

This is contradictory.

If you want to, and you succeed, let us know (maybe describing how in
a wiki page).

If you don't succeed, please wait not until 10.04 will be released by
Ubuntu (Lucid Lynx, at end of April), but until a couple weeks after
the 10.04.1 will be released.

At that point Lucid Lynx will be a solid platform for running a
server, and an LTS platform, supported for 5 years, and the community
will put efforts in supporting it.

-giovanni




> follows:
> g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crti.o: No such file
> or directory
> g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtbeginS.o: No such file or
> directory
> g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtendS.o: No such file or directory
> g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crtn.o: No such file
> or directory
> make[2]: *** [libfreeswitch.la] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch.trunk'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch.trunk'
> make: *** [freeswitch] Error 2
>
> I think this is related to the gcc version; ubuntu 10.04 comes with version
> 4.4.3 rather than 4.4.1.
> Does anyone know how to fix this?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From kees at mroffice.org  Mon Mar 22 01:38:44 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Mon, 22 Mar 2010 21:38:44 +1300
Subject: [Freeswitch-users] Compiling freeswitch on Ubuntu 10.04
In-Reply-To: <7b197bef1003220125x408cc328qfd952addb5280a0b@mail.gmail.com>
References: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>
	<7b197bef1003220125x408cc328qfd952addb5280a0b@mail.gmail.com>
Message-ID: <98d38dcf1003220138g4ae9371ct962155c3db061c69@mail.gmail.com>

It is beta 3, however it is pretty solid and will be released next
month. Besides, I don't want to run a production server on it, I want to try
out fscomm.

Anyway, back to the point: can anybody provide any hints as to how to fix
this?

Thanks,

Kees

On Mon, Mar 22, 2010 at 21:25, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Mon, Mar 22, 2010 at 8:43 AM, Kees Varekamp <kees at mroffice.org> wrote:
> > Hi there,
> > I am trying to compile Freeswitch on Ubuntu 10.04 but am getting stranded
> as
>
> Kees, you're trying to use a non released (alpha, whatever) Ubuntu
> version to build what is intended to be a solid server for telephony.
>
> This is contradictory.
>
> If you want to, and you succeed, let us know (maybe describing how in
> a wiki page).
>
> If you don't succeed, please wait not until 10.04 will be released by
> Ubuntu (Lucid Lynx, at end of April), but until a couple weeks after
> the 10.04.1 will be released.
>
> At that point Lucid Lynx will be a solid platform for running a
> server, and an LTS platform, supported for 5 years, and the community
> will put efforts in supporting it.
>
> -giovanni
>
>
>
>
> > follows:
> > g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crti.o: No such
> file
> > or directory
> > g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtbeginS.o: No such file or
> > directory
> > g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/crtendS.o: No such file or
> directory
> > g++: /usr/lib/gcc/i486-linux-gnu/4.4.1/../../../../lib/crtn.o: No such
> file
> > or directory
> > make[2]: *** [libfreeswitch.la] Error 1
> > make[2]: Leaving directory `/usr/src/freeswitch.trunk'
> > make[1]: *** [all] Error 2
> > make[1]: Leaving directory `/usr/src/freeswitch.trunk'
> > make: *** [freeswitch] Error 2
> >
> > I think this is related to the gcc version; ubuntu 10.04 comes with
> version
> > 4.4.3 rather than 4.4.1.
> > Does anyone know how to fix this?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at microcomaustralia.com.au  Mon Mar 22 01:54:57 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Mon, 22 Mar 2010 19:54:57 +1100
Subject: [Freeswitch-users] freeswitch and android
In-Reply-To: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
References: <3c5cf5261003201753t7bdf770fg5e98d0f33aa46115@mail.gmail.com>
Message-ID: <3c5cf5261003220154w7c486072y7a305c7cc03fa423@mail.gmail.com>

On 21 March 2010 11:53, Brian May <brian at microcomaustralia.com.au> wrote:
> Anybody here got freeswitch working with Android?

Looks like the audio quality issues I had in my test were simply
feedback loops have the phones two close, with one being on speaker
phone. Ooops!

Still doesn't explain why, when a caller rang and I answered I
couldn't hear her and she complained of getting "computer generated
sounds" (whatever that means).

So back to the ringing problem.
-- 
Brian May <brian at microcomaustralia.com.au>


From jingwei.yang at gmail.com  Mon Mar 22 02:13:09 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 22 Mar 2010 17:13:09 +0800
Subject: [Freeswitch-users]  mod_dingaling endless_playback issue
Message-ID: <13529f9d1003220213p11cdeb04l4b05032087ca4f29@mail.gmail.com>

Hello,

I just upgraded to the latest svn (r17063) today and found an issue of
mod_dingaling. I use the command below to originate a call to a gtalk client
and make him hear a piece of endless music:

originate dingaling/gmail.com/jingwei.yang at gmail.com 33333

Here's how the dialplan is defined.

<action application="sleep" data="2000"/>
<action application="info"/>
<action application="endless_playback"
data="customized/to_client_hold_call.mp3"/>

After the call is answered, the client can't hear the music at all and after
30 seconds or so, the call is dropped automatically. I checked this dialplan
with a skypopen client and it worked with no prob (and it used to work in
r16600). Please let me know where goes wrong.

Thanks and best regards,
-Jingwei
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From jingwei.yang at gmail.com  Mon Mar 22 02:20:41 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 22 Mar 2010 17:20:41 +0800
Subject: [Freeswitch-users] mod_dingaling endless_playback issue
In-Reply-To: <13529f9d1003220213p11cdeb04l4b05032087ca4f29@mail.gmail.com>
References: <13529f9d1003220213p11cdeb04l4b05032087ca4f29@mail.gmail.com>
Message-ID: <13529f9d1003220220g691b7a5fwfa58f82a38e32d81@mail.gmail.com>

Sorry, it's my own mistake.The ext-rtp-ip in client.xml wasn't correct.

On Mon, Mar 22, 2010 at 5:13 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hello,
>
> I just upgraded to the latest svn (r17063) today and found an issue of
> mod_dingaling. I use the command below to originate a call to a gtalk client
> and make him hear a piece of endless music:
>
> originate dingaling/gmail.com/jingwei.yang at gmail.com 33333
>
> Here's how the dialplan is defined.
>
> <action application="sleep" data="2000"/>
> <action application="info"/>
> <action application="endless_playback"
> data="customized/to_client_hold_call.mp3"/>
>
> After the call is answered, the client can't hear the music at all and
> after 30 seconds or so, the call is dropped automatically. I checked this
> dialplan with a skypopen client and it worked with no prob (and it used to
> work in r16600). Please let me know where goes wrong.
>
> Thanks and best regards,
> -Jingwei
>
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From abid_freeswitch at live.com  Mon Mar 22 02:32:09 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Mon, 22 Mar 2010 14:32:09 +0500
Subject: [Freeswitch-users] Radius Authentication/Authorization
Message-ID: <COL109-W257A985DAA0C39684BC79EF4270@phx.gbl>


Hi,
I am trying to find out if FreeSwitch supports Radius Authentication/Authorization with free radius or any other radius server. If Yes, How can I configure and If No, what are any alternative workarounds to make it work for prepaid call authentication/authorization.
THanks in advance for help.
Regards
-----------Abid SaleemSr. Product Manager 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From monemran at gmail.com  Mon Mar 22 04:22:31 2010
From: monemran at gmail.com (M.Emran)
Date: Mon, 22 Mar 2010 17:07:31 +0545
Subject: [Freeswitch-users] Radius Authentication/Authorization
In-Reply-To: <COL109-W257A985DAA0C39684BC79EF4270@phx.gbl>
References: <COL109-W257A985DAA0C39684BC79EF4270@phx.gbl>
Message-ID: <b95af1c1003220422v7b9c1888obfe6a350d9a811ad@mail.gmail.com>

You can write it via perl-scripts what we are using.

-- 
Regards
----------
M Emran
Managing Director
E-SOFT BILLING PTE. LTD.
Web: www.e-softbilling.com


On Mon, Mar 22, 2010 at 3:17 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

> Hi,
>
> I am trying to find out if FreeSwitch supports Radius
> Authentication/Authorization with free radius or any other radius server. If
> Yes, How can I configure and If No, what are any alternative workarounds to
> make it work for prepaid call authentication/authorization.
>
> THanks in advance for help.
>
> Regards
>
> -----------
> Abid Saleem
> Sr. Product Manager
>
> ------------------------------
> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at infosecurity.ch  Mon Mar 22 05:34:46 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 13:34:46 +0100
Subject: [Freeswitch-users] Way to *always force* media relay across
	FreeSWITCH
Message-ID: <4BA763E6.9060305@infosecurity.ch>


Hi all,

i am using FreeSWITCH with mobile phones and have problems in FS 
sometimes acting as a media relay, sometimes not.

The mobile phones are behind operator's 3G/2G data networks that are 
"natted" .

The FS just need to act as a SIP server handling:
- Signaling (with SIP/TLS)
- Media relay

NO transcoding is done. (disabled)
Mobile devices use an obfuscated RTP protocol (to bypass VoIP blocking 
by mobile operator) and in order to let FS relay it we use:
inbound-proxy-media=true . That way FS does not even check the RTP 
header but just relay udp packets on agreed ports.

I am having a strong problem, sometimes devices get one-way audio and 
doing debugging we found out that sometimes FS rewrite correctly the SDP 
header but sometimes it does not.

We are using:
<param name="apply-nat-acl" value="nat.auto"/>
<param name="aggressive-nat-detection" value="true"/>

I would like FS to act like Asterisk canreinvite=no that force the media 
to go trough it always rewriting SDP headers and to make the RTP flow 
goes across Asterisk.

I also tried to add this in order to force the rewriting:

      <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>

However given 2 peers sometime i see the status of the peer (sofia 
status profile internal) shown as "Registered(AUTO-NAT)" and sometimes 
as "Registered(TLS)" .

Some times the traffic between the peers is not correctly relayed and 
there is an asymmetric one-way audio situation:
peer-A correctly send RTP traffic to FS that relay it to peer-B
peer-B send RTP traffic directly to peer-B (because FS did not rewrote 
the SDP header related to the target IP of RTP).

 From our basic diagnosis it seems that FS sometimes detect devices as 
"behind NAT" but sometimes it does not.

Is there a way to certainly and precisely force all connections to be 
relayed across the FS like we was doing with "canreinvite=yes" of asterisk?

I just would like a static set of settings without the dynamic methods 
of NAT detection.
Just everything (RTP relay) going trough FS.

Is that possible?

Fabio Pietrosanti
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From lists at infosecurity.ch  Mon Mar 22 05:39:21 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 13:39:21 +0100
Subject: [Freeswitch-users] Unauthenticated users vs activated users
Message-ID: <4BA764F9.8030408@infosecurity.ch>

Hi all,

on my system i completely disabled any kind of users authentication with 
passwords.

However i still need to check if a specific URI is "enabled" to login 
(even if unauthenticated).

So i still need to check if the user can REGISTER or not, checking if 
it's "username" is enabled to login or not.

However when i disable authentication FS does not anymore make xml_curl 
lookup, reasonably, because there's no authentication to be done.

However i need to check in a database if the user (checking only the 
login-name) is authorized to login with user/password.

Is there some workaround and/or particular configuration with which this 
kind of logic could be implemented/enforced or does it require code 
modifications?

If it required introducing a feature to FS i am available to provide 
financial support for consulting to implement such feature.

Waiting for yours
Fabio Pietrosanti


From brian at freeswitch.org  Mon Mar 22 06:05:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 08:05:18 -0500
Subject: [Freeswitch-users] Way to *always force* media relay across
	FreeSWITCH
In-Reply-To: <4BA763E6.9060305@infosecurity.ch>
References: <4BA763E6.9060305@infosecurity.ch>
Message-ID: <381BA1CA-3962-406E-AC16-0281C60D1BB5@freeswitch.org>

It does this by default... you shouldn't need anything to make it behave like that... wipe your configs out and start over... it should just work.

/b

On Mar 22, 2010, at 7:34 AM, Fabio Pietrosanti (naif) wrote:

> I just would like a static set of settings without the dynamic methods of NAT detection.
> Just everything (RTP relay) going trough FS.



From brian at freeswitch.org  Mon Mar 22 06:05:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 08:05:48 -0500
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <4BA764F9.8030408@infosecurity.ch>
References: <4BA764F9.8030408@infosecurity.ch>
Message-ID: <231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>

Please see mod_xml_curl

/b

On Mar 22, 2010, at 7:39 AM, Fabio Pietrosanti (naif) wrote:

> 
> If it required introducing a feature to FS i am available to provide 
> financial support for consulting to implement such feature.



From Russell.Mosemann at cune.org  Mon Mar 22 06:07:40 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Mon, 22 Mar 2010 13:07:40 -0000
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
Message-ID: <20100322130740.19C542704F3@cuneorg-email.cune.pri>

Anthony Minessale <anthony.minessale at gmail.com> said:
> I had a look at the code and I discovered that the zt mod defaults
> to a echo can level of 64 internally. I did a patch to the code
> that should allow it to be explicitly off when you omit that param.

I just tested it, and it works perfectly. Thanks!

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From brian at freeswitch.org  Mon Mar 22 06:21:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 08:21:36 -0500
Subject: [Freeswitch-users] Compiling freeswitch on Ubuntu 10.04
In-Reply-To: <98d38dcf1003220138g4ae9371ct962155c3db061c69@mail.gmail.com>
References: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>
	<7b197bef1003220125x408cc328qfd952addb5280a0b@mail.gmail.com>
	<98d38dcf1003220138g4ae9371ct962155c3db061c69@mail.gmail.com>
Message-ID: <307EEAE2-F14A-4A5A-85B1-1F8FD88DD8CA@freeswitch.org>

Its not so solid if you have a cutting edge compiler... You also failed to provide the logs before the error... Chances are since you decided what to provide to the list and cut out the stuff we really needed to see.  Please include the 100-200 lines previous this error.  Also did you bootstrap this on a different machine and compile the tree then move it?

/b

On Mar 22, 2010, at 3:38 AM, Kees Varekamp wrote:

> It is beta 3, however it is pretty solid and will be released next month. Besides, I don't want to run a production server on it, I want to try out fscomm.
> 
> Anyway, back to the point: can anybody provide any hints as to how to fix this?
> 
> Thanks, 
> 
> Kees



From maciej.aniserowicz at gmail.com  Mon Mar 22 06:36:34 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 22 Mar 2010 05:36:34 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
References: <1268811580808-4748784.post@n2.nabble.com>
	<E2E6F9C7-76FC-40F7-B972-CBD8F6A753AA@freeswitch.org>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
Message-ID: <1269264994788-4777263.post@n2.nabble.com>


Hello again,
Thanks for the reponses, but unfortunately the problem still exists.

This time it happened immediately, so the log in pastebin contains only 2
calls: http://pastebin.freeswitch.org/12514 .

I enclosed 'show channels' output before bridging like you requested.

In my application I wait until ANSWERED and PARK event occur before I do any
processing.

Best regards,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4777263.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kenfulmer at icstechnologysolutions.com  Mon Mar 22 06:49:48 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 22 Mar 2010 08:49:48 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
	<27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>
Message-ID: <008101cac9c6$8a4a8e90$9edfabb0$@com>

I'm not sure I understand. Aren't the sip profiles where I define the ip
address of my internal PBX and / or an upstream carrier? If so, how can I
only use one? 



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Justin B
Newman
Sent: Friday, March 19, 2010 2:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

> larger customer deployments. If we want to create a trunk to an ITSP and a
> trunk to an internal PBX, would we use the external gateway XML files for
> both? We don't plan to point any internal phones to the FreeSwitch box
> (we're using sipX as an internal PBX).

The "internal" and "external" monikers are arbitrary names given to
two SIP profiles.  It's completely possible to run a FreeSWITCH box
with a single profile, named internal, external, or general, or
whatever.  The samples are provided and so-labelled because one of the
biggest problems new users have with PBXs is distinguishing inbound
and outbound calls.  If an inbound call somehow gets into an outbound
dialplan, you can end up with a situation where someone starts using
your PBX to make calls, without your consent.

FreeSWITCH, by permitting association of ports and/or IP addresses
with specific SIP profiles makes it trivial to keep these calls
completely segregated, making it a little harder to do "stupid"
things.

-jbn

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From lists at infosecurity.ch  Mon Mar 22 06:54:00 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 14:54:00 +0100
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>
References: <4BA764F9.8030408@infosecurity.ch>
	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>
Message-ID: <4BA77678.9000803@infosecurity.ch>

Yes,
i am already using mod_xml_curl but mod_xml_curl act only when you 
configure it for a lookup for authenticating users.

I am using it with xml_curl.conf with "directory" and "dialplan" 
succesfull with user authentication .

If i completely disable FS authentication:
internal_auth_calls=false

users can log-in directly without authentication.
But no query to mod_xml_curl is done because authentication is disabled.

However i need to first check if they are *authorized* to log-in without 
authentication and this decision has to be made by querying a database 
(with mod_xml_curl).

I need to check if a users is authorized to login without authentication 
by querying a database with mod_xml_curl.

This does not seems to be possible, already made some discussion on 
#freeswitch channel and David said that it seems to be a feature requests.

Fabio

On 22/03/10 14.05, Brian West wrote:
> Please see mod_xml_curl
>
> /b
>
> On Mar 22, 2010, at 7:39 AM, Fabio Pietrosanti (naif) wrote:
>
>    
>> If it required introducing a feature to FS i am available to provide
>> financial support for consulting to implement such feature.
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From lists at infosecurity.ch  Mon Mar 22 06:55:30 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 14:55:30 +0100
Subject: [Freeswitch-users] Way to *always force* media relay across
	FreeSWITCH
In-Reply-To: <381BA1CA-3962-406E-AC16-0281C60D1BB5@freeswitch.org>
References: <4BA763E6.9060305@infosecurity.ch>
	<381BA1CA-3962-406E-AC16-0281C60D1BB5@freeswitch.org>
Message-ID: <4BA776D2.4010803@infosecurity.ch>

Is possible that the default settings, having some automatic mechanism 
to identify natted vs. non-natted situation (like users having a private 
IP that's not strictly RFC 192.168, 10. o 172.16 but just unallocated 
networkd), make FS does not always detect that a user is behind NAT but 
consider it like if it was outside NAT?

Fabio

On 22/03/10 14.05, Brian West wrote:
> It does this by default... you shouldn't need anything to make it behave like that... wipe your configs out and start over... it should just work.
>
> /b
>
>    


From brian at freeswitch.org  Mon Mar 22 07:00:58 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 09:00:58 -0500
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <4BA77678.9000803@infosecurity.ch>
References: <4BA764F9.8030408@infosecurity.ch>
	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>
	<4BA77678.9000803@infosecurity.ch>
Message-ID: <097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org>

yes turn auth-calls = true.. the hint is to return an empty password if you want to let them in.  So you do all your logic server side.

/b

On Mar 22, 2010, at 8:54 AM, Fabio Pietrosanti (naif) wrote:

> Yes,
> i am already using mod_xml_curl but mod_xml_curl act only when you 
> configure it for a lookup for authenticating users.
> 
> I am using it with xml_curl.conf with "directory" and "dialplan" 
> succesfull with user authentication .
> 
> If i completely disable FS authentication:
> internal_auth_calls=false
> 
> users can log-in directly without authentication.
> But no query to mod_xml_curl is done because authentication is disabled.
> 
> However i need to first check if they are *authorized* to log-in without 
> authentication and this decision has to be made by querying a database 
> (with mod_xml_curl).
> 
> I need to check if a users is authorized to login without authentication 
> by querying a database with mod_xml_curl.
> 
> This does not seems to be possible, already made some discussion on 
> #freeswitch channel and David said that it seems to be a feature requests.
> 
> Fabio



From brian at freeswitch.org  Mon Mar 22 07:01:24 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 09:01:24 -0500
Subject: [Freeswitch-users] Way to *always force* media relay across
	FreeSWITCH
In-Reply-To: <4BA776D2.4010803@infosecurity.ch>
References: <4BA763E6.9060305@infosecurity.ch>
	<381BA1CA-3962-406E-AC16-0281C60D1BB5@freeswitch.org>
	<4BA776D2.4010803@infosecurity.ch>
Message-ID: <95C3BE6F-84C7-4C8B-8F77-C584BA29E23F@freeswitch.org>

Again all media all the time will proxy via FS doesn't matter if its natted or not.

/b

On Mar 22, 2010, at 8:55 AM, Fabio Pietrosanti (naif) wrote:

> Is possible that the default settings, having some automatic mechanism 
> to identify natted vs. non-natted situation (like users having a private 
> IP that's not strictly RFC 192.168, 10. o 172.16 but just unallocated 
> networkd), make FS does not always detect that a user is behind NAT but 
> consider it like if it was outside NAT?
> 
> Fabio



From lists at infosecurity.ch  Mon Mar 22 07:09:06 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 15:09:06 +0100
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org>
References: <4BA764F9.8030408@infosecurity.ch>	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>	<4BA77678.9000803@infosecurity.ch>
	<097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org>
Message-ID: <4BA77A02.8060707@infosecurity.ch>

That's true but the reason for which i am disabling authentication is 
that i am using SIP/TLS over a mobile channel.

Mobile channel is extremely unreliable and the packet loss and high 
delay trigger TCP congestion control systems.

This cause *a lot of problems* increasing the delay for calling 
exponentially.

So i need to reduce the amount of packets (and SIP protocol steps) 
required in order to let the mobile clients to:
a) register
b) INVITE

When authentication is enabled the clients always attempt to login 
without authentication, get a "407" proxy authentication required and 
then try to re-register with authentication using Digest authentication.

The same happen when there is an INVITE.

Removing authentication let me have 50% reduced time and 50% less 
bandwidth in order to let a mobile client register and make a call.

So definitely i need to keep auth-calls=false .

At the same time i need to check if the users logging-in are 
"authorized" to login (without authentication) or not.

Could it be done with FS as-is or does it require a custom 
modification/patches/features in the code?

Fabio Pietrosanti

On 22/03/10 15.00, Brian West wrote:
> yes turn auth-calls = true.. the hint is to return an empty password if you want to let them in.  So you do all your logic server side.
>
> /b
>
> On Mar 22, 2010, at 8:54 AM, Fabio Pietrosanti (naif) wrote:
>
>    
>> Yes,
>> i am already using mod_xml_curl but mod_xml_curl act only when you
>> configure it for a lookup for authenticating users.
>>
>> I am using it with xml_curl.conf with "directory" and "dialplan"
>> succesfull with user authentication .
>>
>> If i completely disable FS authentication:
>> internal_auth_calls=false
>>
>> users can log-in directly without authentication.
>> But no query to mod_xml_curl is done because authentication is disabled.
>>
>> However i need to first check if they are *authorized* to log-in without
>> authentication and this decision has to be made by querying a database
>> (with mod_xml_curl).
>>
>> I need to check if a users is authorized to login without authentication
>> by querying a database with mod_xml_curl.
>>
>> This does not seems to be possible, already made some discussion on
>> #freeswitch channel and David said that it seems to be a feature requests.
>>
>> Fabio
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From brian at freeswitch.org  Mon Mar 22 07:27:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 09:27:48 -0500
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <4BA77A02.8060707@infosecurity.ch>
References: <4BA764F9.8030408@infosecurity.ch>	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>	<4BA77678.9000803@infosecurity.ch>
	<097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org>
	<4BA77A02.8060707@infosecurity.ch>
Message-ID: <CD7426D6-5526-44EA-831E-66CED88BAF2E@freeswitch.org>

Please pay close attention this time.

On Mar 22, 2010, at 9:09 AM, Fabio Pietrosanti (naif) wrote:

> When authentication is enabled the clients always attempt to login 
> without authentication, get a "407" proxy authentication required and 
> then try to re-register with authentication using Digest authentication.
> 
> The same happen when there is an INVITE.
> 
> Removing authentication let me have 50% reduced time and 50% less 
> bandwidth in order to let a mobile client register and make a call.

You clearly do not get what I'm saying.  I have told you what to do and you still don't listen.

> So definitely i need to keep auth-calls=false .

No you want it to true.  If you don't listen to me this time then I give up.

The reason you want it set to true is then you CAN do the directory lookup.  Just return an empty password. The 401/407 will never take place and it gives you a way to let them in or not depending on what you want.

> At the same time i need to check if the users logging-in are 
> "authorized" to login (without authentication) or not.

See Above.

> Could it be done with FS as-is or does it require a custom 
> modification/patches/features in the code?

See Above.

> 
> Fabio Pietrosanti

/b



From justin at ejtown.org  Mon Mar 22 07:29:18 2010
From: justin at ejtown.org (Justin B Newman)
Date: Mon, 22 Mar 2010 10:29:18 -0400
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <008101cac9c6$8a4a8e90$9edfabb0$@com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>
	<27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>
	<008101cac9c6$8a4a8e90$9edfabb0$@com>
Message-ID: <27b0d5bc1003220729l56c60e60p14387122e7b8defe@mail.gmail.com>

On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> I'm not sure I understand. Aren't the sip profiles where I define the ip
> address of my internal PBX and / or an upstream carrier? If so, how can I
> only use one?

A SIP profile associates an IP address and port with a series of
settings and with a series of gateways.  In so much as the gateways
are defined properly, I'm not aware of an inherent need to segregate
them into two SIP profiles.

That said, having incoming and outgoing calls going through the same
SIP profile significantly increases the opportunity for error, which
could result in security vulnerabilities.

-jbn


From gmaruzz at celliax.org  Mon Mar 22 07:32:27 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 22 Mar 2010 15:32:27 +0100
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <CD7426D6-5526-44EA-831E-66CED88BAF2E@freeswitch.org>
References: <4BA764F9.8030408@infosecurity.ch>
	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org> 
	<4BA77678.9000803@infosecurity.ch>
	<097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org> 
	<4BA77A02.8060707@infosecurity.ch>
	<CD7426D6-5526-44EA-831E-66CED88BAF2E@freeswitch.org>
Message-ID: <7b197bef1003220732t3648da92ycc104911a3f11ba0@mail.gmail.com>

bkw wishlist:

http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0


On Mon, Mar 22, 2010 at 3:27 PM, Brian West <brian at freeswitch.org> wrote:
> Please pay close attention this time.
>
> On Mar 22, 2010, at 9:09 AM, Fabio Pietrosanti (naif) wrote:
>
>> When authentication is enabled the clients always attempt to login
>> without authentication, get a "407" proxy authentication required and
>> then try to re-register with authentication using Digest authentication.
>>
>> The same happen when there is an INVITE.
>>
>> Removing authentication let me have 50% reduced time and 50% less
>> bandwidth in order to let a mobile client register and make a call.
>
> You clearly do not get what I'm saying. ?I have told you what to do and you still don't listen.
>
>> So definitely i need to keep auth-calls=false .
>
> No you want it to true. ?If you don't listen to me this time then I give up.
>
> The reason you want it set to true is then you CAN do the directory lookup. ?Just return an empty password. The 401/407 will never take place and it gives you a way to let them in or not depending on what you want.
>
>> At the same time i need to check if the users logging-in are
>> "authorized" to login (without authentication) or not.
>
> See Above.
>
>> Could it be done with FS as-is or does it require a custom
>> modification/patches/features in the code?
>
> See Above.
>
>>
>> Fabio Pietrosanti
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lists at infosecurity.ch  Mon Mar 22 08:06:52 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 16:06:52 +0100
Subject: [Freeswitch-users] Unauthenticated users vs activated users
In-Reply-To: <CD7426D6-5526-44EA-831E-66CED88BAF2E@freeswitch.org>
References: <4BA764F9.8030408@infosecurity.ch>	<231C0564-A153-4A07-9450-10087D0DF339@freeswitch.org>	<4BA77678.9000803@infosecurity.ch>	<097604F2-3577-4F5E-ACF2-9930591CB4BC@freeswitch.org>	<4BA77A02.8060707@infosecurity.ch>
	<CD7426D6-5526-44EA-831E-66CED88BAF2E@freeswitch.org>
Message-ID: <4BA7878C.8010407@infosecurity.ch>

On 22/03/10 15.27, Brian West wrote:
> Please pay close attention this time.
>    
Reading very closely, let me answer to see my setup.
>> So definitely i need to keep auth-calls=false .
>>      
> No you want it to true.  If you don't listen to me this time then I give up.
>
> The reason you want it set to true is then you CAN do the directory lookup.  Just return an empty password. The 401/407 will never take place and it gives you a way to let them in or not depending on what you want.
>    

I verified that it does not works properly that way (at least with my setup)
1 - Client -> REGISTER(unauth) -> Server
2 - Client <- 407/401 (send digest) <- Server
3 - Client -> REGISTER(digest auth) -> Server
4 - Client <- OK <- Server

The FreeSWITCH configured with xml_curl.conf as follow:
<binding name="directory">
<param name="gateway-url" value="http://XXX/directory" 
bindings="directory"/>
<param name="auth-scheme" value="digest"/>
</binding>
<binding name="dialplan">
<param name="gateway-url" value="http://XXX/dialplan" bindings="dialplan"/>
</binding>

I verified with tcpdump in one hand and siptrace on the other one and 
with this setup FS make the xml_curl requests only AFTER the step "3", 
so after the unauth requests already failed.

Maybe i am doing something wrong at the configuration, but FS does not 
make any kind of lookup on step "1" of the sip registration process.

I need FS to make a lookup to a database on step "1" of authentication 
process, so i can check that THAT sip UA is enabled to login without 
authentication.

Some ideas on how to tweak this situation?

Fabio Pietrosanti


From kenfulmer at icstechnologysolutions.com  Mon Mar 22 08:55:55 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 22 Mar 2010 10:55:55 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <27b0d5bc1003220729l56c60e60p14387122e7b8defe@mail.gmail.com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>	<27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>	<008101cac9c6$8a4a8e90$9edfabb0$@com>
	<27b0d5bc1003220729l56c60e60p14387122e7b8defe@mail.gmail.com>
Message-ID: <00a001cac9d8$28607750$792165f0$@com>

Ok, thanks. I think I'm beginning to understand a little more clearly. 

Ken

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Justin B
Newman
Sent: Monday, March 22, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> I'm not sure I understand. Aren't the sip profiles where I define the ip
> address of my internal PBX and / or an upstream carrier? If so, how can I
> only use one?

A SIP profile associates an IP address and port with a series of
settings and with a series of gateways.  In so much as the gateways
are defined properly, I'm not aware of an inherent need to segregate
them into two SIP profiles.

That said, having incoming and outgoing calls going through the same
SIP profile significantly increases the opportunity for error, which
could result in security vulnerabilities.

-jbn

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From Prometheus001 at gmx.net  Mon Mar 22 09:01:01 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 22 Mar 2010 17:01:01 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>
References: <4B9F71AD.7070008@gmx.net>
	<2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>
Message-ID: <4BA7943D.9030001@gmx.net>

I tried this, but only with half success

after the phones responds with "do not disturb"
(Variables from Info application
variable_proto_specific_hangup_cause: [sip:480]
variable_sip_hangup_phrase: [Do Not Disturb])

I do process the following:
<action application="bridge"
data="loopback/hangup_causes_${sip_hangup_phrase}"/>

In the following dialplan I process:
<extension name="hangup-causes">
<action application="info" data=""/>
<condition field="destination_number" expression="hangup_causes_Do Not
Disturb">
<action application="sleep" data="1000"/>
<action application="set" data="sip_ignore_remote_cause=true"/>
<!-- <action application="respond" data="486"/> -->
<action application="hangup" data="USER_BUSY"/>
</condition>
</extension>
Freeswitch sends a 480 Message with
Reason: Q.850;cause=17;text="USER_BUSY"

This is a first step, but I need a 486 message as the PSTN still reponds
"temporary unavailable" instead of "busy". Also "respond" with "486"
does not work, as commented in the dialplan.

It seems that the 480 message cannot be overriden in this case.
Freeswitch insists on using 480
mod_sofia.c:352 sofia/internal/203 at mydomain Overriding SIP cause 486
with 480 from the other leg

Any other idea to achieve a 486 message?

Best regards
Peter

Leon de Rooij schrieb:
> Hi,
>
> Don't know if it helps, but the inline attribute can only be set on  
> actions, not on conditions.
>
> Also, the hangup application is not allowed to be run inline:
> http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_dptools/mod_dptools.c?r=HEAD#l3047
> (it has no SAF_ROUTING_EXEC flag)
>
> Perhaps you can try to transfer to another context directly after your  
> bridge ? (where that second context contains an extension with  
> conditions matching on variable_originate_disposition)
>
> kind regards,
>
> Leon
>
>
> On Mar 16, 2010, at 12:55 PM, Peter P GMX wrote:
>
>   
>> I have the folowing scenario:
>>  A Snom phone has activated DND (do not disturb)
>>  When the phone is called, it returns "480 Do Not Disturb" and then
>> "404 Not Found".
>>  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
>>  The PSTN then (wrongly) announces "this number is not listed".
>>
>> My question: How can we overcome this and announce back a different
>> message to PSTN, but only on NO_USER_RESPONSE.
>> I tried to continue on bridge and then this after the bridge command  
>> the
>> following:
>>      <condition field="variable_originate_disposition"
>> expression="NO_USER_RESPONSE" inline="true">
>>        <action application="hangup" data="USER_BUSY"/>
>>      </condition>
>>
>> But the installed Freeswith does not support the "inline" yet and we
>> will not be able to update Freeswitch soon.
>>
>> So my question:
>> Is there any other way to achieve this?
>>
>> Best regards
>> Peter
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From kenfulmer at icstechnologysolutions.com  Mon Mar 22 09:05:14 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 22 Mar 2010 11:05:14 -0500
Subject: [Freeswitch-users] Newbie Questions
In-Reply-To: <27b0d5bc1003220729l56c60e60p14387122e7b8defe@mail.gmail.com>
References: <00a701cac776$bf147de0$3d3d79a0$@com>	<27b0d5bc1003191217h1dfc0685y5afaf16cf587c3f6@mail.gmail.com>	<008101cac9c6$8a4a8e90$9edfabb0$@com>
	<27b0d5bc1003220729l56c60e60p14387122e7b8defe@mail.gmail.com>
Message-ID: <00a101cac9d9$7595d3c0$60c17b40$@com>

Sorry for the repeat message.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Justin B
Newman
Sent: Monday, March 22, 2010 9:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Newbie Questions

On Mon, Mar 22, 2010 at 9:49 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:
> I'm not sure I understand. Aren't the sip profiles where I define the ip
> address of my internal PBX and / or an upstream carrier? If so, how can I
> only use one?

A SIP profile associates an IP address and port with a series of
settings and with a series of gateways.  In so much as the gateways
are defined properly, I'm not aware of an inherent need to segregate
them into two SIP profiles.

That said, having incoming and outgoing calls going through the same
SIP profile significantly increases the opportunity for error, which
could result in security vulnerabilities.

-jbn

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Mar 22 09:43:01 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 11:43:01 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1269264994788-4777263.post@n2.nabble.com>
References: <1268811580808-4748784.post@n2.nabble.com>
	<1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<1269264994788-4777263.post@n2.nabble.com>
Message-ID: <191c3a031003220943j7d67d87cv805a17c9874413a6@mail.gmail.com>

did you add this to both dial strings too (requires latest trunk)
{uuid_bridge_continue_on_cancel=true}




On Mon, Mar 22, 2010 at 8:36 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello again,
> Thanks for the reponses, but unfortunately the problem still exists.
>
> This time it happened immediately, so the log in pastebin contains only 2
> calls: http://pastebin.freeswitch.org/12514 .
>
> I enclosed 'show channels' output before bridging like you requested.
>
> In my application I wait until ANSWERED and PARK event occur before I do
> any
> processing.
>
> Best regards,
> Maciej Aniserowicz
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4777263.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From tculjaga at gmail.com  Mon Mar 22 09:45:09 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 22 Mar 2010 17:45:09 +0100
Subject: [Freeswitch-users] Radius Authentication/Authorization
In-Reply-To: <b95af1c1003220422v7b9c1888obfe6a350d9a811ad@mail.gmail.com>
References: <COL109-W257A985DAA0C39684BC79EF4270@phx.gbl>
	<b95af1c1003220422v7b9c1888obfe6a350d9a811ad@mail.gmail.com>
Message-ID: <65d96fc81003220945h3e78bb68v388037108ee2833d@mail.gmail.com>

no need for Perl :)

http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_rad_auth/mod_rad_auth.c?r=17037

add this to your modules.conf "applications/mod_rad_auth"


and just make mod_rad_auth-install this will do the job!

after that you need to configure it correctly... i didn't have time to put
everything on wiki but will do it this week.



T.


On Mon, Mar 22, 2010 at 12:22 PM, M.Emran <monemran at gmail.com> wrote:

> You can write it via perl-scripts what we are using.
>
> --
> Regards
> ----------
> M Emran
> Managing Director
> E-SOFT BILLING PTE. LTD.
> Web: www.e-softbilling.com
>
>
> On Mon, Mar 22, 2010 at 3:17 PM, Abid Saleem <abid_freeswitch at live.com>wrote:
>
>> Hi,
>>
>> I am trying to find out if FreeSwitch supports Radius
>> Authentication/Authorization with free radius or any other radius server. If
>> Yes, How can I configure and If No, what are any alternative workarounds to
>> make it work for prepaid call authentication/authorization.
>>
>> THanks in advance for help.
>>
>> Regards
>>
>> -----------
>> Abid Saleem
>> Sr. Product Manager
>>
>> ------------------------------
>> Hotmail: Powerful Free email with security by Microsoft. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar 22 09:59:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Mar 2010 09:59:55 -0700
Subject: [Freeswitch-users] How to upgrade configuration files when
	upgrading FreeSwitch
In-Reply-To: <d361ca7b1003202254i4df16131x9b5cac6e10851750@mail.gmail.com>
References: <d361ca7b1003202254i4df16131x9b5cac6e10851750@mail.gmail.com>
Message-ID: <87f2f3b91003220959n71769e7amb61c127a5de522b2@mail.gmail.com>

On Sat, Mar 20, 2010 at 10:54 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com
> wrote:

> Hello,
>
>   Once a week I upgrade our production system by retreiving
> freeswitch-1.0.5-latest.tar.gz, building and installing it. This updates all
> the binaries, but I am left with the current config files (which is good, as
> some of them contains locally edited data). The problem is how to detect
> (and merge) changes in the configuration files, like new parameters. I can
> do manual diffs, but it is a time consuming task.
>
> What is the BCP of doing such an upgrade?
>

I would do automated diffs from a script. In fact, it might be easiest to
diff the new default configs against the existing default configs so that
you only see what was changed in the past week. That would be less noisy
than diffing against your modified config files.

How's your bash scripting skills? :)
-MC
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From maciej.aniserowicz at gmail.com  Mon Mar 22 10:04:28 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 22 Mar 2010 09:04:28 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003220943j7d67d87cv805a17c9874413a6@mail.gmail.com>
References: <1268846266197-4751694.post@n2.nabble.com>
	<3D73C745-8D87-4AE2-9508-83E2797F7F98@freeswitch.org>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<1269264994788-4777263.post@n2.nabble.com>
	<191c3a031003220943j7d67d87cv805a17c9874413a6@mail.gmail.com>
Message-ID: <1269277468028-4779041.post@n2.nabble.com>


No, I did not. I understood that this var was only to recover from incorrect
scenario which I manually introduced by misplacing the "ignore_early_media"
declaration... and that I should completely avoid this situation instead of
just using such dirty fix.
I'll update to trunk and let you know if if this additional var gets the job
done.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4779041.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lists at infosecurity.ch  Mon Mar 22 10:04:24 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 18:04:24 +0100
Subject: [Freeswitch-users] Way to *always force* media relay across
	FreeSWITCH
In-Reply-To: <4BA776D2.4010803@infosecurity.ch>
References: <4BA763E6.9060305@infosecurity.ch>
	<381BA1CA-3962-406E-AC16-0281C60D1BB5@freeswitch.org>
	<4BA776D2.4010803@infosecurity.ch>
Message-ID: <4BA7A318.5000404@infosecurity.ch>

That's a situation in the FS debug log when media relay is in action:
[INFO] switch_rtp.c:2005 Auto Changing port from 10.220.24.40:4002 to 
213.230.130.210:14166
[INFO] switch_rtp.c:2005 Auto Changing port from 192.168.11.207:4000 to 
93.35.50.222:4000

That's a situation in the FS debug log when media relay activate only on 
one leg of the call:
[INFO] switch_rtp.c:2005 Auto Changing port from 192.168.11.207:4001 to 
93.35.50.222:4001

I will do deeply trying to dig into the code of auto-nat why sometimes 
it does not act as a media relay for one leg.

I am really getting crazy (also with the default configuration)

Fabio

On 22/03/10 14.55, Fabio Pietrosanti (naif) wrote:
> Is possible that the default settings, having some automatic mechanism 
> to identify natted vs. non-natted situation (like users having a 
> private IP that's not strictly RFC 192.168, 10. o 172.16 but just 
> unallocated networkd), make FS does not always detect that a user is 
> behind NAT but consider it like if it was outside NAT?
>
> Fabio
>
> On 22/03/10 14.05, Brian West wrote:
>> It does this by default... you shouldn't need anything to make it 
>> behave like that... wipe your configs out and start over... it should 
>> just work.
>>
>> /b
>>



From rupa at rupa.com  Mon Mar 22 10:18:34 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 22 Mar 2010 12:18:34 -0500
Subject: [Freeswitch-users] How to upgrade configuration files when
	upgrading FreeSwitch
In-Reply-To: <87f2f3b91003220959n71769e7amb61c127a5de522b2@mail.gmail.com>
References: <d361ca7b1003202254i4df16131x9b5cac6e10851750@mail.gmail.com>
	<87f2f3b91003220959n71769e7amb61c127a5de522b2@mail.gmail.com>
Message-ID: <b39273531003221018wd37e3c9o6955ceb9a9e73fbe@mail.gmail.com>

Look at how your distro does it.  Debian has a utility called ucf for
managing the 3way diff between old default, new default, and currently
installed.  It does require some setup though and isn't necessarily obvious.
 There are other tools and other distros for this.

You can also look at using something like git or svn.  Read up on how to
track a "vendor branch" using your tool of choice.

On Mon, Mar 22, 2010 at 11:59 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Sat, Mar 20, 2010 at 10:54 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Hello,
>>
>>   Once a week I upgrade our production system by retreiving
>> freeswitch-1.0.5-latest.tar.gz, building and installing it. This updates all
>> the binaries, but I am left with the current config files (which is good, as
>> some of them contains locally edited data). The problem is how to detect
>> (and merge) changes in the configuration files, like new parameters. I can
>> do manual diffs, but it is a time consuming task.
>>
>> What is the BCP of doing such an upgrade?
>>
>
> I would do automated diffs from a script. In fact, it might be easiest to
> diff the new default configs against the existing default configs so that
> you only see what was changed in the past week. That would be less noisy
> than diffing against your modified config files.
>
> How's your bash scripting skills? :)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From fraserredmond at gmail.com  Mon Mar 22 10:23:10 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 22 Mar 2010 17:23:10 +0000
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <4BA5EF7A.9010200@sip-communicator.org>
References: <4BA5EF7A.9010200@sip-communicator.org>
Message-ID: <c32ee1cf1003221023r2ef7dd4cra18dcf8ccb4d66ed@mail.gmail.com>

Emil, can you just clarify: If we have our pool of licenses set up on the
server, is it then just a matter of enabling g729 in the code? (Or are
additional licenses needed on the client somehow?)

Cheers,
Fraser




On Sun, Mar 21, 2010 at 10:05 AM, Emil Ivov <emcho at sip-communicator.org>wrote:

>
> Yes we do have it, and it was implemented as a part of our participation
> in GSoC 2009. People can use it perfectly legally as long as they buy
> the corresponding licenses (just as with any other client).
>
> Currently use of the codec is disabled by default and whoever needs it
> has to enable it in the code, after acquiring the necessary license.
>
> Cheers,
> Emil
>
> >
> > /b
> >
> > On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:
> >
> >> I believe sip communicator does handle g729. It has a strong
> >> community and a plugin model (based on osgi) that makes it easily
> >> (ok, kinda...) customizable. Of course the
> >> FSComm/givesomemoneytotheproject option is the best one. In any
> >> case: http://sip-communicator.org/
> >>
> >> Ciao, R
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Mar 22 10:33:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 12:33:23 -0500
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <1269277468028-4779041.post@n2.nabble.com>
References: <1268846266197-4751694.post@n2.nabble.com>
	<1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<1269264994788-4777263.post@n2.nabble.com>
	<191c3a031003220943j7d67d87cv805a17c9874413a6@mail.gmail.com>
	<1269277468028-4779041.post@n2.nabble.com>
Message-ID: <191c3a031003221033k534961b1ua4f4d93ea0ebd3dc@mail.gmail.com>

update again to 17069 as well
if you still get the problem and you really want to find out why

add debugging code to switch_ivr_wait_for_answer switch_ivr_originate.c:738
to find out which code patch causes it to return false.


On Mon, Mar 22, 2010 at 12:04 PM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> No, I did not. I understood that this var was only to recover from
> incorrect
> scenario which I manually introduced by misplacing the "ignore_early_media"
> declaration... and that I should completely avoid this situation instead of
> just using such dirty fix.
> I'll update to trunk and let you know if if this additional var gets the
> job
> done.
> --
> View this message in context:
> http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4779041.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at redbonez.net  Mon Mar 22 10:40:30 2010
From: lists at redbonez.net (Adam Ford)
Date: Mon, 22 Mar 2010 11:40:30 -0600
Subject: [Freeswitch-users] Migrating voicemail to new freeswitch instance
Message-ID: <002801cac9e6$c57a0610$506e1230$@net>

How would I migrate people's custom recorded voicemail greetings to a new
install of FreeSWITCH? I finally got upgraded from 1.0.4 to SVN trunk.

 

Thanks,

-Adam

 

 

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From brian at freeswitch.org  Mon Mar 22 10:47:40 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 12:47:40 -0500
Subject: [Freeswitch-users] Migrating voicemail to new freeswitch
	instance
In-Reply-To: <002801cac9e6$c57a0610$506e1230$@net>
References: <002801cac9e6$c57a0610$506e1230$@net>
Message-ID: <0D32AD38-3AC6-47C7-92D6-046D32604C28@freeswitch.org>

That didn't change anything.  So move the database you should be fine.

/b

On Mar 22, 2010, at 12:40 PM, Adam Ford wrote:

> How would I migrate people?s custom recorded voicemail greetings to a new install of FreeSWITCH? I finally got upgraded from 1.0.4 to SVN trunk.
>  
> Thanks,
> -Adam

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From steveu at coppice.org  Mon Mar 22 10:48:29 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Mar 2010 01:48:29 +0800
Subject: [Freeswitch-users] Open source softphone that can handle g729
In-Reply-To: <c32ee1cf1003221023r2ef7dd4cra18dcf8ccb4d66ed@mail.gmail.com>
References: <4BA5EF7A.9010200@sip-communicator.org>
	<c32ee1cf1003221023r2ef7dd4cra18dcf8ccb4d66ed@mail.gmail.com>
Message-ID: <4BA7AD6D.9070201@coppice.org>

On 03/23/2010 01:23 AM, Fraser Redmond wrote:
> Emil, can you just clarify: If we have our pool of licenses set up on 
> the server, is it then just a matter of enabling g729 in the code? (Or 
> are additional licenses needed on the client somehow?)
>
> Cheers,
> Fraser
The conditions for G.729 patent licencing say each point in the chain 
which transcodes must be licenced. You need a licence for the client and 
a licence for the server, unless the server is operating in pass-through 
mode. Licences are not transferrable. You can't buy, say, a licence for 
the Freeswitch G.729 codec, but actually run a copy of the Java code 
from SIP Communicator. The products must be individually licenced. Those 
are the rules set by the licencing pool.
>
> On Sun, Mar 21, 2010 at 10:05 AM, Emil Ivov 
> <emcho at sip-communicator.org <mailto:emcho at sip-communicator.org>> wrote:
>
>
>     Yes we do have it, and it was implemented as a part of our
>     participation
>     in GSoC 2009. People can use it perfectly legally as long as they buy
>     the corresponding licenses (just as with any other client).
>
>     Currently use of the codec is disabled by default and whoever needs it
>     has to enable it in the code, after acquiring the necessary license.
>
>     Cheers,
>     Emil
>
>     >
>     > /b
>     >
>     > On Mar 20, 2010, at 8:24 AM, Raffaele P. Guidi wrote:
>     >
>     >> I believe sip communicator does handle g729. It has a strong
>     >> community and a plugin model (based on osgi) that makes it easily
>     >> (ok, kinda...) customizable. Of course the
>     >> FSComm/givesomemoneytotheproject option is the best one. In any
>     >> case: http://sip-communicator.org/
>     >>
>     >> Ciao, R
>

Steve



From anthony.minessale at gmail.com  Mon Mar 22 11:08:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 13:08:34 -0500
Subject: [Freeswitch-users] Now Hiring for CudaTEL UI Programmer
Message-ID: <191c3a031003221108u1c502219qef68969a6656229@mail.gmail.com>

As most people know, the CudaTEL uses FreeSWITCH internally as the telephony
engine.
We're now looking to expand the development team:

We're looking for a UI programmer with jquery experience.

See:

http://jobs.jsninja.com/job/1598/jqueryjs-ui-developer-at-barracuda-networks/

Anyone interested,  please apply to this forum or email jobs at freeswitch.org

Thanks.


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Mon Mar 22 11:16:40 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 13:16:40 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100322130740.19C542704F3@cuneorg-email.cune.pri>
References: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
	<20100322130740.19C542704F3@cuneorg-email.cune.pri>
Message-ID: <191c3a031003221116g2490d31ai5f89e60e46429290@mail.gmail.com>

great,

Now all the time I wasted on asterisk has finally paid off, I knew I
remembered something from back then!


On Mon, Mar 22, 2010 at 8:07 AM, <Russell.Mosemann at cune.org> wrote:

> Anthony Minessale <anthony.minessale at gmail.com> said:
> > I had a look at the code and I discovered that the zt mod defaults
> > to a echo can level of 64 internally. I did a patch to the code
> > that should allow it to be explicitly off when you omit that param.
>
> I just tested it, and it works perfectly. Thanks!
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Mon Mar 22 12:02:39 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 22 Mar 2010 20:02:39 +0100
Subject: [Freeswitch-users] sip-force-contact: which to choose?
Message-ID: <4BA7BECF.6060209@infosecurity.ch>

Hi all,

i read in latest source code of FS (of yesterday) that for the handling 
of sip-force-contact now there is a new option:

NDLB-connectile-dysfunction-2.0

Which are the improvement of NDLB-connectile-dysfunction-2.0 respect to 
NDLB-connectile-dysfunction?

I read that the 2.0 also make the users registered with "AUTO-NAT 2.0".

Fabio


From msc at freeswitch.org  Mon Mar 22 12:19:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Mar 2010 12:19:16 -0700
Subject: [Freeswitch-users] How to find the number of live calls for a
	specific dialplan
In-Reply-To: <B91AFBDD83964B558D84B4D859794C1F@sharad>
References: <B91AFBDD83964B558D84B4D859794C1F@sharad>
Message-ID: <87f2f3b91003221219r3e47db84r22d82571975849e1@mail.gmail.com>

On Fri, Mar 19, 2010 at 10:41 PM, sharad <sharad at coraltele.com> wrote:

>  Hi All
>
> If there any way to find out the number of live calls for a specific
> dialplan.
>
You might want to investigate mod_limit. The module name is a bit misleading
in that it does more than merely limit the use of resources. It can be used
for much more.
-MC
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From infos at madovsky.org  Mon Mar 22 12:20:30 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 22 Mar 2010 15:20:30 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <B720C7D1F11C468AAA3720B96EF35326@MOBILEE1705>
Message-ID: <4BBCFCF9845C4BCF8F080B720F623216@MOBILEE1705>

so when I use a public IP to register a phone,
there s a rtp timeout from line 853 of switch_rtp.c

I looked at the code line 853 below :

-------------------------------------------------------
        old_sock = rtp_session->sock_input;
        rtp_session->sock_input = new_sock;
        new_sock = NULL;

        if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER) || switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK)) {
                switch_socket_opt_set(rtp_session->sock_input, SWITCH_SO_NONBLOCK, TRUE);
                switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_NOBLOCK);
        }

        switch_socket_create_pollset(&rtp_session->read_pollfd, rtp_session->sock_input, SWITCH_POLLIN | SWITCH_POLLERR, rtp_session->pool);

        status = SWITCH_STATUS_SUCCESS;
        *err = "Success";
        switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_IO);

  done:

        if (new_sock) {
                switch_socket_close(new_sock);
        }

        if (old_sock) {
                switch_socket_close(old_sock);
        }

        if (rtp_session->ready != 1) {
                WRITE_DEC(rtp_session);
                READ_DEC(rtp_session);
        }

        return status;

-----------------------------------

it seems that old_sock is the actual sock that it needs to use since there is no auto changing to do ?
So does it mean maybe the old_sock is yet existing because there is no NAT ?

Thanks




----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 22, 2010 2:31 AM
Subject: Re: [Freeswitch-users] rtp timeout yet


>I deactivated iptables, no router, no nat,
> only 2 public IP, 2 softphones connected on the same FS which is on a public
> IP to.
> default config, result -> rtp timeout
> crazy that's not working since I can make 2 softphones behind a firewall
> work....
> 
> ok that's enough for me tonight ! ;)
> 
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, March 22, 2010 1:55 AM
> Subject: Re: [Freeswitch-users] rtp timeout yet
> 
> 
>>I replace public ip by these fake one...
>>
>> ----- Original Message ----- 
>> From: "Jason White" <jason at jasonjgw.net>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 22, 2010 1:49 AM
>> Subject: Re: [Freeswitch-users] rtp timeout yet
>>
>>
>>> Madovsky <infos at madovsky.org> wrote:
>>>> result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060
>>>>
>>>> 10.10.10.132 = softphone1 registered from public IP
>>>>
>>>> 11.11.11.= softphone2 registered from public IP
>>>>
>>>> 10.10.10.138 = FS
>>>
>>> Those are private addresses; there's a NAT device involved somewhere, and
>>> that's likely to be where your problems are.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Mon Mar 22 14:42:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Mar 2010 14:42:53 -0700
Subject: [Freeswitch-users] auth incoming calls
In-Reply-To: <3c5cf5261003220124g20f065au4f60196bdbbe82c0@mail.gmail.com>
References: <3c5cf5261003220124g20f065au4f60196bdbbe82c0@mail.gmail.com>
Message-ID: <87f2f3b91003221442k7686ad48x36d830d98421f720@mail.gmail.com>

Fire up the console and check the debug output when the call first comes in.
Look for any references to an acl because that's where you'll see if the
call is authing or not...

-MC

On Mon, Mar 22, 2010 at 1:24 AM, Brian May
<brian at microcomaustralia.com.au>wrote:

> Hello,
>
> In debugging that previous problem I noticed another problem.
>
> How do I get freeswitch to authenticate incomming calls? As far as I
> can tell it is not working:
>
> 483.970655 192.168.87.13 -> 202.173.153.98 SIP/SDP Request: INVITE
> sip:9999 at 202.173.153.98 <sip%3A9999 at 202.173.153.98>, with session
> description
> 483.972989 202.173.153.98 -> 192.168.87.13 SIP Status: 100 Trying
> 484.180018 202.173.153.98 -> 192.168.87.13 SIP/SDP Status: 200 OK,
> with session description
> 484.208932 192.168.87.13 -> 202.173.153.98 SIP Request: ACK
> sip:9999 at 202.173.153.98:5060;transport=udp
> 490.634429 192.168.87.13 -> 202.173.153.98 SIP Request: BYE
> sip:9999 at 202.173.153.98:5060;transport=udp
> 490.637862 202.173.153.98 -> 192.168.87.13 SIP Status: 200 OK
>
> Yet the setting is appears to be enabled:
>
> <param name="auth-calls" value="true"/>
>
> --
> Brian May <brian at microcomaustralia.com.au>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraserredmond at gmail.com  Mon Mar 22 14:47:59 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 22 Mar 2010 21:47:59 +0000
Subject: [Freeswitch-users] Bridge or Intercept?
Message-ID: <c32ee1cf1003221447pb455604u3a66dc6dacfe50ac@mail.gmail.com>

I want to call an extn, then when they pick up connect them to another
existing channel - later I'll run this via event_socket, but for now I'm
doing it in javascript (I want to do database updates before and after
attempting the connection.)

As far as I can see, I have two options, and I'm interested in which is
better, and more importantly - how to get either of them working!


1) Create the new session, then intercept the existing channel:

var NewSession = new
Session('{ignore_early_media=true,hangup_after_bridge=false}user/
12610 at domainname.com');
NewSession.execute('intercept', 'cafc3449-9000-47bc-8b28-257748d7ec40');

The newSession phone rings and on pickup both channels get hung up. The
ExistingChannel gives a call status in the CLI of [DESTINATION_OUT_OF_ORDER]


2) Create the new session, and create a session object for the existing
session and bridge them:

var NewSession = new
Session('{ignore_early_media=true,hangup_after_bridge=false}user/
12610 at domainname.com');
var ExistingSession = new Session('cafc3449-9000-47bc-8b28-257748d7ec40');
bridge(ExistingSession, NewSession);

(This has a debug message saying "BRIDGE THREAD DONE", but not much else
that seems unusual.)

The sound files that are playing in ExistingSession continue, and after
about 10sec A notice appears in the CLI indicating a session.hungup event
has fired, but both phones remain on the hook and hanging up either doesn't
affect the other.


Cheers,
Fraser
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From anthony.minessale at gmail.com  Mon Mar 22 15:00:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Mar 2010 17:00:07 -0500
Subject: [Freeswitch-users] Bridge or Intercept?
In-Reply-To: <c32ee1cf1003221447pb455604u3a66dc6dacfe50ac@mail.gmail.com>
References: <c32ee1cf1003221447pb455604u3a66dc6dacfe50ac@mail.gmail.com>
Message-ID: <191c3a031003221500n131e2fcfj37af6eb1abc98f6f@mail.gmail.com>

how about no script and 1 command

originate ignore_early_media=true,hangup_after_bridge=false}user/
12610 at domainname.com intercept:cafc3449-9000-47bc-8b28-257748d7ec40 inline

(Providing that uuid is of a channel that is ready to bridge)



On Mon, Mar 22, 2010 at 4:47 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I want to call an extn, then when they pick up connect them to another
> existing channel - later I'll run this via event_socket, but for now I'm
> doing it in javascript (I want to do database updates before and after
> attempting the connection.)
>
> As far as I can see, I have two options, and I'm interested in which is
> better, and more importantly - how to get either of them working!
>
>
> 1) Create the new session, then intercept the existing channel:
>
> var NewSession = new
> Session('{ignore_early_media=true,hangup_after_bridge=false}user/
> 12610 at domainname.com');
> NewSession.execute('intercept', 'cafc3449-9000-47bc-8b28-257748d7ec40');
>
> The newSession phone rings and on pickup both channels get hung up. The
> ExistingChannel gives a call status in the CLI of [DESTINATION_OUT_OF_ORDER]
>
>
> 2) Create the new session, and create a session object for the existing
> session and bridge them:
>
> var NewSession = new
> Session('{ignore_early_media=true,hangup_after_bridge=false}user/
> 12610 at domainname.com');
> var ExistingSession = new Session('cafc3449-9000-47bc-8b28-257748d7ec40');
> bridge(ExistingSession, NewSession);
>
> (This has a debug message saying "BRIDGE THREAD DONE", but not much else
> that seems unusual.)
>
> The sound files that are playing in ExistingSession continue, and after
> about 10sec A notice appears in the CLI indicating a session.hungup event
> has fired, but both phones remain on the hook and hanging up either doesn't
> affect the other.
>
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From msc at freeswitch.org  Mon Mar 22 15:03:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Mar 2010 15:03:04 -0700
Subject: [Freeswitch-users] Monitoring Registrations/Users
In-Reply-To: <2534666.101269222324377.JavaMail.root@zimbra1.spigotsystems.com>
References: <33146668.81269222287052.JavaMail.root@zimbra1.spigotsystems.com>
	<2534666.101269222324377.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <87f2f3b91003221503p67d9bd69y728d48a64a1f57e6@mail.gmail.com>

You can monitor in various ways, but a simple way to start out is by using
either a script with ESL or just doing: fs_cli -x "sofia status profile
internal" and parsing the results to look for whichever SIP registrations
are of interest.

As far as finding this useful, I'm sure there are some who would find it
useful in at least a few cases. However, my guess is that they would be more
interested in know why the registrations are getting dropped in the first
place...

-MC

On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore <
dave.redmore at spigotsystems.com> wrote:

> Been working with Freeswitch to implement a simple T38 proxy to provide
> outbound faxing to some customers.  Works very well, on initial testing,
> using the proxy-media option.  Nice NAT detection/SBC functionality "out of
> the box".  Generally very easy to get my head around and get it to do what I
> wanted,
>
> One bit of functionality I've been looking for in a softswitch for a while,
> and have never seen, is to somehow monitor the status of users/registrations
> and alert somehow if there is a problem.  In my situation, we have connected
> some customers with VoIP phone service for their small offices.  Typically,
> just a simple ATA that gives them 1-4 trunks and a couple of DIDs.  The
> problem has always been that if their ATA drops it's registration or is not
> available, no one knows it until a customer tells them through some other
> means that their phones aren't working.  This is a huge problem, obviously,
> and happens more than I would like.  It seems like it wouldn't be too
> difficult to create some sort of monitoring function which can send an
> Invite packet every little while and make sure the UA responds with a
> 100/Trying and then disconnect.  Basically polling any devices that I deem
> as needing to always be available and sending an email or something if one
> isn't.  Does this seem useful to anyone else?  Has anyone seen anything like
> this out there?
>
> Thanks,
>
> Dave Redmore
> Spigot Networks, Inc.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Mon Mar 22 15:32:35 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 22 Mar 2010 22:32:35 +0000
Subject: [Freeswitch-users] Bridge or Intercept?
In-Reply-To: <191c3a031003221500n131e2fcfj37af6eb1abc98f6f@mail.gmail.com>
References: <c32ee1cf1003221447pb455604u3a66dc6dacfe50ac@mail.gmail.com> 
	<191c3a031003221500n131e2fcfj37af6eb1abc98f6f@mail.gmail.com>
Message-ID: <c32ee1cf1003221532u2a8a51d9tc201da6a79e23cba@mail.gmail.com>

Thanks Anthony! That works great. I read about inlines today, but didn't
really get it, I'll have to go back and read it again.

I think this is the main (only) problem I've seen with FreeSwitch - there's
so much you can do, and often in just one line, but to get that one line you
often have to ask the creator! :-)

Cheers,
Fraser




On Mon, Mar 22, 2010 at 10:00 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> how about no script and 1 command
>
> originate ignore_early_media=true,hangup_after_bridge=false}user/
> 12610 at domainname.com intercept:cafc3449-9000-47bc-8b28-257748d7ec40 inline
>
> (Providing that uuid is of a channel that is ready to bridge)
>
>
>
> On Mon, Mar 22, 2010 at 4:47 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:
>
>> I want to call an extn, then when they pick up connect them to another
>> existing channel - later I'll run this via event_socket, but for now I'm
>> doing it in javascript (I want to do database updates before and after
>> attempting the connection.)
>>
>> As far as I can see, I have two options, and I'm interested in which is
>> better, and more importantly - how to get either of them working!
>>
>>
>> 1) Create the new session, then intercept the existing channel:
>>
>> var NewSession = new
>> Session('{ignore_early_media=true,hangup_after_bridge=false}user/
>> 12610 at domainname.com');
>> NewSession.execute('intercept', 'cafc3449-9000-47bc-8b28-257748d7ec40');
>>
>> The newSession phone rings and on pickup both channels get hung up. The
>> ExistingChannel gives a call status in the CLI of [DESTINATION_OUT_OF_ORDER]
>>
>>
>> 2) Create the new session, and create a session object for the existing
>> session and bridge them:
>>
>> var NewSession = new
>> Session('{ignore_early_media=true,hangup_after_bridge=false}user/
>> 12610 at domainname.com');
>> var ExistingSession = new Session('cafc3449-9000-47bc-8b28-257748d7ec40');
>> bridge(ExistingSession, NewSession);
>>
>> (This has a debug message saying "BRIDGE THREAD DONE", but not much else
>> that seems unusual.)
>>
>> The sound files that are playing in ExistingSession continue, and after
>> about 10sec A notice appears in the CLI indicating a session.hungup event
>> has fired, but both phones remain on the hook and hanging up either doesn't
>> affect the other.
>>
>>
>> Cheers,
>> Fraser
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Mar 22 15:40:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Mar 2010 15:40:50 -0700
Subject: [Freeswitch-users] Bridge or Intercept?
In-Reply-To: <c32ee1cf1003221532u2a8a51d9tc201da6a79e23cba@mail.gmail.com>
References: <c32ee1cf1003221447pb455604u3a66dc6dacfe50ac@mail.gmail.com>
	<191c3a031003221500n131e2fcfj37af6eb1abc98f6f@mail.gmail.com>
	<c32ee1cf1003221532u2a8a51d9tc201da6a79e23cba@mail.gmail.com>
Message-ID: <87f2f3b91003221540t732998b8n856a41e66fe5ff34@mail.gmail.com>

On Mon, Mar 22, 2010 at 3:32 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks Anthony! That works great. I read about inlines today, but didn't
> really get it, I'll have to go back and read it again.
>
> I think this is the main (only) problem I've seen with FreeSwitch - there's
> so much you can do, and often in just one line, but to get that one line you
> often have to ask the creator! :-)
>

Just try that with Asterisk! :P
-MC
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From dave.redmore at spigotsystems.com  Mon Mar 22 16:01:03 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Mon, 22 Mar 2010 18:01:03 -0500 (CDT)
Subject: [Freeswitch-users] Monitoring Registrations/Users
In-Reply-To: <12217984.511269298856722.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <14020578.531269298863488.JavaMail.root@zimbra1.spigotsystems.com>

Thanks for the reply. I think that for me, the reason that the registrations are getting dropped or calls are getting dropped is because we are running across "best effort:" Internet connections - DSL and Cable - and we are using off-the-shelf ATAs from Grandstream and Linksys. So, stuff happens and we lose calls - but if I could Poll those far ends, I can be more proactive when that happens. If I could send a SIP packet of some sort every minute and make sure I get a response, then I know those devices are ready to receive a call and if they aren't I can proactively find out why. 

Dave 

----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Monday, March 22, 2010 5:03:04 PM GMT -06:00 US/Canada Central 
Subject: Re: [Freeswitch-users] Monitoring Registrations/Users 

You can monitor in various ways, but a simple way to start out is by using either a script with ESL or just doing: fs_cli -x "sofia status profile internal" and parsing the results to look for whichever SIP registrations are of interest. 

As far as finding this useful, I'm sure there are some who would find it useful in at least a few cases. However, my guess is that they would be more interested in know why the registrations are getting dropped in the first place... 

-MC 


On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore < dave.redmore at spigotsystems.com > wrote: 




Been working with Freeswitch to implement a simple T38 proxy to provide outbound faxing to some customers. Works very well, on initial testing, using the proxy-media option. Nice NAT detection/SBC functionality "out of the box". Generally very easy to get my head around and get it to do what I wanted, 

One bit of functionality I've been looking for in a softswitch for a while, and have never seen, is to somehow monitor the status of users/registrations and alert somehow if there is a problem. In my situation, we have connected some customers with VoIP phone service for their small offices. Typically, just a simple ATA that gives them 1-4 trunks and a couple of DIDs. The problem has always been that if their ATA drops it's registration or is not available, no one knows it until a customer tells them through some other means that their phones aren't working. This is a huge problem, obviously, and happens more than I would like. It seems like it wouldn't be too difficult to create some sort of monitoring function which can send an Invite packet every little while and make sure the UA responds with a 100/Trying and then disconnect. Basically polling any devices that I deem as needing to always be available and sending an email or something if one isn't. Does this seem useful to anyone else? Has anyone seen anything like this out there? 

Thanks, 

Dave Redmore 
Spigot Networks, Inc. 

_______________________________________________ 
FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 



_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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From gkuri at ieee.org  Mon Mar 22 16:16:12 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Mon, 22 Mar 2010 16:16:12 -0700
Subject: [Freeswitch-users] Monitoring Registrations/Users
In-Reply-To: <14020578.531269298863488.JavaMail.root@zimbra1.spigotsystems.com>
References: <12217984.511269298856722.JavaMail.root@zimbra1.spigotsystems.com>
	<14020578.531269298863488.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <8b1c9cda1003221616i544ba297y9d6f2ec1a45ce94c@mail.gmail.com>

Try and use "unregister-on-options-fail" and "nat-options-ping" ?
Write a little app that connects to the Event Socket and wait for a
device to be unregistered as a trigger for doing whatever you want (ie
send an email, snmp trap, etc.).

    http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-options-fail

Cheers,
Gabe

On Mon, Mar 22, 2010 at 4:01 PM, Dave  Redmore
<dave.redmore at spigotsystems.com> wrote:
> Thanks for the reply.? I think that for me, the reason that the
> registrations are getting dropped or calls are getting dropped is because we
> are running across "best effort:" Internet connections - DSL and Cable - and
> we are using off-the-shelf ATAs from Grandstream and Linksys.? So, stuff
> happens and we lose calls - but if I could Poll those far ends, I can be
> more proactive when that happens.? If I could send a SIP packet of some sort
> every minute and make sure I get a response, then I know those devices are
> ready to receive a call and if they aren't I can proactively find out why.
>
> Dave
>
> ----- Original Message -----
> From: "Michael Collins" <msc at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, March 22, 2010 5:03:04 PM GMT -06:00 US/Canada Central
> Subject: Re: [Freeswitch-users] Monitoring Registrations/Users
>
> You can monitor in various ways, but a simple way to start out is by using
> either a script with ESL or just doing: fs_cli -x "sofia status profile
> internal" and parsing the results to look for whichever SIP registrations
> are of interest.
>
> As far as finding this useful, I'm sure there are some who would find it
> useful in at least a few cases. However, my guess is that they would be more
> interested in know why the registrations are getting dropped in the first
> place...
>
> -MC
>
> On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore
> <dave.redmore at spigotsystems.com> wrote:
>>
>> Been working with Freeswitch to implement a simple T38 proxy to provide
>> outbound faxing to some customers.? Works very well, on initial testing,
>> using the proxy-media option.? Nice NAT detection/SBC functionality "out of
>> the box".? Generally very easy to get my head around and get it to do what I
>> wanted,
>>
>> One bit of functionality I've been looking for in a softswitch for a
>> while, and have never seen, is to somehow monitor the status of
>> users/registrations and alert somehow if there is a problem.? In my
>> situation, we have connected some customers with VoIP phone service for
>> their small offices.? Typically, just a simple ATA that gives them 1-4
>> trunks and a couple of DIDs.? The problem has always been that if their ATA
>> drops it's registration or is not available, no one knows it until a
>> customer tells them through some other means that their phones aren't
>> working.? This is a huge problem, obviously, and happens more than I would
>> like.? It seems like it wouldn't be too difficult to create some sort of
>> monitoring function which can send an Invite packet every little while and
>> make sure the UA responds with a 100/Trying and then disconnect.? Basically
>> polling any devices that I deem as needing to always be available and
>> sending an email or something if one isn't.? Does this seem useful to anyone
>> else?? Has anyone seen anything like this out there?
>>
>> Thanks,
>>
>> Dave Redmore
>> Spigot Networks, Inc.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________ FreeSWITCH-users mailing
> list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dave.redmore at spigotsystems.com  Mon Mar 22 16:34:03 2010
From: dave.redmore at spigotsystems.com (Dave Redmore)
Date: Mon, 22 Mar 2010 18:34:03 -0500 (CDT)
Subject: [Freeswitch-users] Monitoring Registrations/Users
In-Reply-To: <1620541.631269300823335.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <30958517.651269300843711.JavaMail.root@zimbra1.spigotsystems.com>

Nice - This is what I was looking for. So, Freeswitch sends an options Ping to the endpoints, using the nat-options-ping option and then will unregister if an endpoint doesn't respond. Then I can alert on an unregistered device. Sounds perfect. Thanks. Will experiment with that. 

Dave 

----- Original Message ----- 
From: "Gabriel Kuri" <gkuri at ieee.org> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Monday, March 22, 2010 6:16:12 PM GMT -06:00 US/Canada Central 
Subject: Re: [Freeswitch-users] Monitoring Registrations/Users 

Try and use "unregister-on-options-fail" and "nat-options-ping" ? 
Write a little app that connects to the Event Socket and wait for a 
device to be unregistered as a trigger for doing whatever you want (ie 
send an email, snmp trap, etc.). 

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#unregister-on-options-fail 

Cheers, 
Gabe 

On Mon, Mar 22, 2010 at 4:01 PM, Dave Redmore 
<dave.redmore at spigotsystems.com> wrote: 
> Thanks for the reply. I think that for me, the reason that the 
> registrations are getting dropped or calls are getting dropped is because we 
> are running across "best effort:" Internet connections - DSL and Cable - and 
> we are using off-the-shelf ATAs from Grandstream and Linksys. So, stuff 
> happens and we lose calls - but if I could Poll those far ends, I can be 
> more proactive when that happens. If I could send a SIP packet of some sort 
> every minute and make sure I get a response, then I know those devices are 
> ready to receive a call and if they aren't I can proactively find out why. 
> 
> Dave 
> 
> ----- Original Message ----- 
> From: "Michael Collins" <msc at freeswitch.org> 
> To: freeswitch-users at lists.freeswitch.org 
> Sent: Monday, March 22, 2010 5:03:04 PM GMT -06:00 US/Canada Central 
> Subject: Re: [Freeswitch-users] Monitoring Registrations/Users 
> 
> You can monitor in various ways, but a simple way to start out is by using 
> either a script with ESL or just doing: fs_cli -x "sofia status profile 
> internal" and parsing the results to look for whichever SIP registrations 
> are of interest. 
> 
> As far as finding this useful, I'm sure there are some who would find it 
> useful in at least a few cases. However, my guess is that they would be more 
> interested in know why the registrations are getting dropped in the first 
> place... 
> 
> -MC 
> 
> On Sun, Mar 21, 2010 at 6:45 PM, Dave Redmore 
> <dave.redmore at spigotsystems.com> wrote: 
>> 
>> Been working with Freeswitch to implement a simple T38 proxy to provide 
>> outbound faxing to some customers. Works very well, on initial testing, 
>> using the proxy-media option. Nice NAT detection/SBC functionality "out of 
>> the box". Generally very easy to get my head around and get it to do what I 
>> wanted, 
>> 
>> One bit of functionality I've been looking for in a softswitch for a 
>> while, and have never seen, is to somehow monitor the status of 
>> users/registrations and alert somehow if there is a problem. In my 
>> situation, we have connected some customers with VoIP phone service for 
>> their small offices. Typically, just a simple ATA that gives them 1-4 
>> trunks and a couple of DIDs. The problem has always been that if their ATA 
>> drops it's registration or is not available, no one knows it until a 
>> customer tells them through some other means that their phones aren't 
>> working. This is a huge problem, obviously, and happens more than I would 
>> like. It seems like it wouldn't be too difficult to create some sort of 
>> monitoring function which can send an Invite packet every little while and 
>> make sure the UA responds with a 100/Trying and then disconnect. Basically 
>> polling any devices that I deem as needing to always be available and 
>> sending an email or something if one isn't. Does this seem useful to anyone 
>> else? Has anyone seen anything like this out there? 
>> 
>> Thanks, 
>> 
>> Dave Redmore 
>> Spigot Networks, Inc. 
>> 
>> _______________________________________________ 
>> FreeSWITCH-users mailing list 
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>> http://www.freeswitch.org 
>> 
> 
> 
> _______________________________________________ FreeSWITCH-users mailing 
> list FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 
> _______________________________________________ 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
> http://www.freeswitch.org 
> 
> 

_______________________________________________ 
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FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From kees at mroffice.org  Mon Mar 22 17:37:34 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Tue, 23 Mar 2010 13:37:34 +1300
Subject: [Freeswitch-users] Compiling freeswitch on Ubuntu 10.04
In-Reply-To: <307EEAE2-F14A-4A5A-85B1-1F8FD88DD8CA@freeswitch.org>
References: <98d38dcf1003220043p427a8bbdq6e4673c807ad6212@mail.gmail.com>
	<7b197bef1003220125x408cc328qfd952addb5280a0b@mail.gmail.com>
	<98d38dcf1003220138g4ae9371ct962155c3db061c69@mail.gmail.com>
	<307EEAE2-F14A-4A5A-85B1-1F8FD88DD8CA@freeswitch.org>
Message-ID: <98d38dcf1003221737l25af71b7vbeb1e5b59c2e54b@mail.gmail.com>

OK that solved it! I did bootstrap it on another machine.

Recompiled everything according to
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and now it's running
fine on ubuntu 10.04.

Many thanks for the suggestion,

Kees

On Tue, Mar 23, 2010 at 02:21, Brian West <brian at freeswitch.org> wrote:

> Its not so solid if you have a cutting edge compiler... You also failed to
> provide the logs before the error... Chances are since you decided what to
> provide to the list and cut out the stuff we really needed to see.  Please
> include the 100-200 lines previous this error.  Also did you bootstrap this
> on a different machine and compile the tree then move it?
>
> /b
>
> On Mar 22, 2010, at 3:38 AM, Kees Varekamp wrote:
>
> > It is beta 3, however it is pretty solid and will be released next month.
> Besides, I don't want to run a production server on it, I want to try out
> fscomm.
> >
> > Anyway, back to the point: can anybody provide any hints as to how to fix
> this?
> >
> > Thanks,
> >
> > Kees
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jcasale at activenetwerx.com  Mon Mar 22 20:39:21 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 23 Mar 2010 03:39:21 +0000
Subject: [Freeswitch-users] Compile issue with openzap
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>

I just fetched freeswitch-1.0.5-latest.tar.bz2 right now and enabled
openzap in modules.conf and compiled fs, but the openzap bits never
made it in?

I have done it before on this system (centos 5.3 x64) w/o issue, any
known issues atm?

Thanks!
jlc


From brian at freeswitch.org  Mon Mar 22 20:48:40 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Mar 2010 22:48:40 -0500
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
Message-ID: <530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>

Every day... happen to have the output of any errors you had during the compile?

/b

On Mar 22, 2010, at 10:39 PM, Joseph L. Casale wrote:

> I have done it before on this system (centos 5.3 x64) w/o issue, any
> known issues atm?



From jzamora at corvus.com.ve  Mon Mar 22 20:47:58 2010
From: jzamora at corvus.com.ve (=?ISO-8859-1?Q?Jos=E9_A_Zamora?=)
Date: Tue, 23 Mar 2010 23:17:58 +1930
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <20100322130740.19C542704F3@cuneorg-email.cune.pri>
References: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
	<20100322130740.19C542704F3@cuneorg-email.cune.pri>
Message-ID: <dbaae9f61003222047v10c84550v50d98b645d61d046@mail.gmail.com>

hello.

You could write the steps used to finally solve the problem?

2010/3/23 <Russell.Mosemann at cune.org>

> Anthony Minessale <anthony.minessale at gmail.com> said:
> > I had a look at the code and I discovered that the zt mod defaults
> > to a echo can level of 64 internally. I did a patch to the code
> > that should allow it to be explicitly off when you omit that param.
>
> I just tested it, and it works perfectly. Thanks!
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason at jasonjgw.net  Mon Mar 22 20:56:19 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Mar 2010 14:56:19 +1100
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
Message-ID: <20100323035619.GA11860@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Every day... happen to have the output of any errors you had during the
> compile?

And I'm also inclined to ask whether you enabled the module before building
FreeSWITCH.



From jcasale at activenetwerx.com  Mon Mar 22 21:01:30 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 23 Mar 2010 04:01:30 +0000
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E48C8@prato.activenetwerx.local>

>Every day... happen to have the output of any errors you had during the compile?
>
>/b

I have the fs dir saved and the buffer from the terminal saved to a text file.
I have pastebin'ed the freeswitch-1.0.5-20100321-0400/libs/openzap/config.log
at http://pastebin.freeswitch.org/12522 if that helps.

I didn't see anything in the saved buffer from the terminal I was compiling in.

Thanks!
jlc


From infos at madovsky.org  Mon Mar 22 23:47:22 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 02:47:22 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <BF89D404049740658F0B636B508894AC@MOBILEE1705>

to be sure I'm silly I downgraded from trunk 17001
and rtp timeout disappeared.

Regards

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 22, 2010 3:20 PM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  so when I use a public IP to register a phone,
  there s a rtp timeout from line 853 of switch_rtp.c

  I looked at the code line 853 below :

  -------------------------------------------------------
          old_sock = rtp_session->sock_input;
          rtp_session->sock_input = new_sock;
          new_sock = NULL;

          if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER) || switch_test_flag(rtp_session, SWITCH_RTP_FLAG_NOBLOCK)) {
                  switch_socket_opt_set(rtp_session->sock_input, SWITCH_SO_NONBLOCK, TRUE);
                  switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_NOBLOCK);
          }

          switch_socket_create_pollset(&rtp_session->read_pollfd, rtp_session->sock_input, SWITCH_POLLIN | SWITCH_POLLERR, rtp_session->pool);

          status = SWITCH_STATUS_SUCCESS;
          *err = "Success";
          switch_set_flag_locked(rtp_session, SWITCH_RTP_FLAG_IO);

    done:

          if (new_sock) {
                  switch_socket_close(new_sock);
          }

          if (old_sock) {
                  switch_socket_close(old_sock);
          }

          if (rtp_session->ready != 1) {
                  WRITE_DEC(rtp_session);
                  READ_DEC(rtp_session);
          }

          return status;

  -----------------------------------

  it seems that old_sock is the actual sock that it needs to use since there is no auto changing to do ?
  So does it mean maybe the old_sock is yet existing because there is no NAT ?

  Thanks




  ----- Original Message ----- 
  From: "Madovsky" <infos at madovsky.org>
  To: <freeswitch-users at lists.freeswitch.org>
  Sent: Monday, March 22, 2010 2:31 AM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  >I deactivated iptables, no router, no nat,
  > only 2 public IP, 2 softphones connected on the same FS which is on a public
  > IP to.
  > default config, result -> rtp timeout
  > crazy that's not working since I can make 2 softphones behind a firewall
  > work....
  > 
  > ok that's enough for me tonight ! ;)
  > 
  > ----- Original Message ----- 
  > From: "Madovsky" <infos at madovsky.org>
  > To: <freeswitch-users at lists.freeswitch.org>
  > Sent: Monday, March 22, 2010 1:55 AM
  > Subject: Re: [Freeswitch-users] rtp timeout yet
  > 
  > 
  >>I replace public ip by these fake one...
  >>
  >> ----- Original Message ----- 
  >> From: "Jason White" <jason at jasonjgw.net>
  >> To: <freeswitch-users at lists.freeswitch.org>
  >> Sent: Monday, March 22, 2010 1:49 AM
  >> Subject: Re: [Freeswitch-users] rtp timeout yet
  >>
  >>
  >>> Madovsky <infos at madovsky.org> wrote:
  >>>> result with tcpdump -ennqti br0 -T rtp -vvv dst port 5060
  >>>>
  >>>> 10.10.10.132 = softphone1 registered from public IP
  >>>>
  >>>> 11.11.11.= softphone2 registered from public IP
  >>>>
  >>>> 10.10.10.138 = FS
  >>>
  >>> Those are private addresses; there's a NAT device involved somewhere, and
  >>> that's likely to be where your problems are.
  >>>
  >>>
  >>> _______________________________________________
  >>> FreeSWITCH-users mailing list
  >>> FreeSWITCH-users at lists.freeswitch.org
  >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  >>> http://www.freeswitch.org 
  > 
  > 
  > _______________________________________________
  > FreeSWITCH-users mailing list
  > FreeSWITCH-users at lists.freeswitch.org
  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  > http://www.freeswitch.org 
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From yehavi.bourvine at gmail.com  Tue Mar 23 00:19:09 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 23 Mar 2010 09:19:09 +0200
Subject: [Freeswitch-users] FreeSwitch & Vegastream TLS interoperability
	problems?
Message-ID: <d361ca7b1003230019w65b54f7cwbd0c158a86442570@mail.gmail.com>

Hello,

  We have a Vegastream ATA adapter working fine with FreeSwitch as long as
we do not enable TLS. Once TLS is enabled, I see the following behavior:


   - The adapter creates a TLS tunnel with Freeswitch and registers itself.
   However, I see one difference in the registration between this adapter and
   other adapters/phones:
    - The VegaStream adapter is registered as:

      Contact:          "user" <sips:80632 at 10.64.1.6:5061>
      Status:           Registered(TLS)(unknown) EXP(2010-03-23 08:50:26)
      -

      Other TLS phones register as:
      Contact:          "user" <sips:80631 at 10.64.0.8:5061;transport=tls>
      Status:           Registered(TLS)(unknown) EXP(2010-03-23 08:37:35)
      -

      What I see is the difference in the "contact" field in which the
      transport=tls clause is missing.
      -

   When a user on the ATA dials out, the request is sent via TLS and all is
   fine.
   -

   When someone calls an ATA extension, Freeswitch sends the request to the
   ATA via SIP/UDP and not via SIP/TLS

 I've contacted the manufacturer who claims that according to RFC5630 it
should send the request via TLS as it is registered as TLS. I have a feeling
that the erronous behaviour is because they do not append "transport=tls" to
the contact field, although they claim it is not required.



Who is right and who is wrong?



                      Thanks! __Yehavi:
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From jason at jasonjgw.net  Tue Mar 23 01:32:08 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 23 Mar 2010 19:32:08 +1100
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
Message-ID: <20100323083208.GA12866@jdc.jasonjgw.net>

Madovsky <infos at madovsky.org> wrote:
> to be sure I'm silly I downgraded from trunk 17001
> and rtp timeout disappeared.

Try the latest trunk. If the problem still occurs, capture SIP traces and
submit a bug report, making sure the configurations are identical.



From helmut.kuper at ewetel.de  Tue Mar 23 01:37:23 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 23 Mar 2010 09:37:23 +0100
Subject: [Freeswitch-users] Question about TOS/DSCP
Message-ID: <4BA87DC3.2010004@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

is there a way in FS to set the TOS/DSCP values directly in FS instead
of using iptables as described in FS wiki?

regards
helmut
-----BEGIN PGP SIGNATURE-----
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Cv2gtKHQPYyeelW4mxwFpmg=
=AJAv
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From maciej.aniserowicz at gmail.com  Tue Mar 23 02:28:30 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 23 Mar 2010 01:28:30 -0800 (PST)
Subject: [Freeswitch-users] Call gets hanged up with ORIGINATOR_CANCEL
	cause
In-Reply-To: <191c3a031003221033k534961b1ua4f4d93ea0ebd3dc@mail.gmail.com>
References: <1268848740857-4751927.post@n2.nabble.com>
	<191c3a031003171131g4ac25daake1eb7f62400616d5@mail.gmail.com>
	<1268900370911-4755257.post@n2.nabble.com>
	<191c3a031003181156g73eab976r22f8489cb5426af2@mail.gmail.com>
	<191c3a031003181203l3d2f9b8cjd1e58a9288ae4856@mail.gmail.com>
	<191c3a031003181208x26077d92g5d0c8e65c280c315@mail.gmail.com>
	<1269264994788-4777263.post@n2.nabble.com>
	<191c3a031003220943j7d67d87cv805a17c9874413a6@mail.gmail.com>
	<1269277468028-4779041.post@n2.nabble.com>
	<191c3a031003221033k534961b1ua4f4d93ea0ebd3dc@mail.gmail.com>
Message-ID: <1269336510774-4783162.post@n2.nabble.com>


Now, with uuid_bridge_continue_on_cancel set to true, it works fine - no more
hangups. Setting this var is still necessary though, 17069 does not work
without it.

Thanks a lot for your help.
-- 
View this message in context: http://n2.nabble.com/Call-gets-hanged-up-with-ORIGINATOR-CANCEL-cause-tp4735736p4783162.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From andy at fabulous4.co.uk  Tue Mar 23 03:28:48 2010
From: andy at fabulous4.co.uk (Andy)
Date: Tue, 23 Mar 2010 10:28:48 -0000
Subject: [Freeswitch-users] recoding idividual contributions to a conference
	call
Message-ID: <C5A8C13D259D46BB933846A3EF28AFCF@D810>

Hi,
 
Sorry I couldn't find this in the doucmentation. If I have 3 people in a
conference call, can I record each of their contriubutions into 3 seperate
files? 
 
Many thanks
Andy
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From Russell.Mosemann at cune.org  Tue Mar 23 04:10:13 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Tue, 23 Mar 2010 06:10:13 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <dbaae9f61003222047v10c84550v50d98b645d61d046@mail.gmail.com>
References: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com><20100322130740.19C542704F3@cuneorg-email.cune.pri>
	<dbaae9f61003222047v10c84550v50d98b645d61d046@mail.gmail.com>
Message-ID: <5C2D0CB33CC846DDAFC6A0C3ECA877CE@cune.pri>

Jos? A Zamora asked:

> You could write the steps used to finally solve the problem?

Sure. :-)

1. Comment with ";" or remove "echo_cancel_level => 64" in zt.conf.
2. make current

The problem was no audio in either direction when echo cancelling was enabled in DAHDI (version 2.2.1). The problem occurred, because open_zap was setting the echo cancel level, even when echo_cancel_level was commented in zt.conf. Anthony's change turns off echo_cancel_level, if not specified in zt.conf. That allows echo cancelling in DAHDI to work.

--
Russell Mosemann




From jcasale at activenetwerx.com  Tue Mar 23 04:44:20 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 23 Mar 2010 11:44:20 +0000
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <20100323035619.GA11860@jdc.jasonjgw.net>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>

>And I'm also inclined to ask whether you enabled the module before building
>FreeSWITCH.

Yup.


From helmut.kuper at ewetel.de  Tue Mar 23 05:35:56 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 23 Mar 2010 13:35:56 +0100
Subject: [Freeswitch-users] Question about TOS/DSCP
In-Reply-To: <4BA87DC3.2010004@ewetel.de>
References: <4BA87DC3.2010004@ewetel.de>
Message-ID: <4BA8B5AC.30209@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I was able to set it hardcoded in sofia.c of mod_sofia.

Maybe we can get this as a configurable parameter in sofia.conf?


regards
Helmut

On 23.03.2010 09:37, Helmut Kuper wrote:
> Hello,
> 
> is there a way in FS to set the TOS/DSCP values directly in FS instead
> of using iptables as described in FS wiki?
> 
> regards
> helmut

-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

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QJM7CRCDPhctAuIt4nGdrDU=
=p1QB
-----END PGP SIGNATURE-----


From brian at freeswitch.org  Tue Mar 23 06:57:14 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 08:57:14 -0500
Subject: [Freeswitch-users] FreeSwitch & Vegastream TLS interoperability
	problems?
In-Reply-To: <d361ca7b1003230019w65b54f7cwbd0c158a86442570@mail.gmail.com>
References: <d361ca7b1003230019w65b54f7cwbd0c158a86442570@mail.gmail.com>
Message-ID: <755C8A7E-6B78-47D4-AF80-224CB0EA2395@freeswitch.org>

Every endpoint I have tested that works correctly will append the transport=tls so we have a clue what transport to use to call them back.

/b

On Mar 23, 2010, at 2:19 AM, Yehavi Bourvine wrote:

> I've contacted the manufacturer who claims that according to RFC5630 it should send the request via TLS as it is registered as TLS. I have a feeling that the erronous behaviour is because they do not append "transport=tls" to the contact field, although they claim it is not required.

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From brian at freeswitch.org  Tue Mar 23 07:00:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 09:00:26 -0500
Subject: [Freeswitch-users] Question about TOS/DSCP
In-Reply-To: <4BA8B5AC.30209@ewetel.de>
References: <4BA87DC3.2010004@ewetel.de> <4BA8B5AC.30209@ewetel.de>
Message-ID: <648899B0-74DB-4340-85E8-3FD4B0342477@freeswitch.org>

the iptables way is better for now.  I thought we exposed those in Sofia but thats just for Signaling and not Media.... I would use the iptables way.

/b

On Mar 23, 2010, at 7:35 AM, Helmut Kuper wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Hi,
> 
> I was able to set it hardcoded in sofia.c of mod_sofia.
> 
> Maybe we can get this as a configurable parameter in sofia.conf?
> 
> 
> regards
> Helmut
> 
> On 23.03.2010 09:37, Helmut Kuper wrote:
>> Hello,
>> 
>> is there a way in FS to set the TOS/DSCP values directly in FS instead
>> of using iptables as described in FS wiki?
>> 
>> regards
>> helmut
> 
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
> 
> iD8DBQFLqLWs4tZeNddg3dwRAnAcAKCYek1W8v6QJmceaBnOe9a3QtpuCgCfdhTo
> QJM7CRCDPhctAuIt4nGdrDU=
> =p1QB
> -----END PGP SIGNATURE-----
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Mar 23 07:00:56 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 09:00:56 -0500
Subject: [Freeswitch-users] recoding idividual contributions to a
	conference call
In-Reply-To: <C5A8C13D259D46BB933846A3EF28AFCF@D810>
References: <C5A8C13D259D46BB933846A3EF28AFCF@D810>
Message-ID: <74ACD32C-1615-4FCD-8542-C4589B779074@freeswitch.org>

record each person not the conference.

/b

On Mar 23, 2010, at 5:28 AM, Andy wrote:

> Sorry I couldn't find this in the doucmentation. If I have 3 people in a conference call, can I record each of their contriubutions into 3 seperate files?
>  

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From brian at freeswitch.org  Tue Mar 23 07:01:31 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 09:01:31 -0500
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
	<CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
Message-ID: <486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>

What is the output of the build when you type make?

/b

On Mar 23, 2010, at 6:44 AM, Joseph L. Casale wrote:

>> And I'm also inclined to ask whether you enabled the module before building
>> FreeSWITCH.
> 
> Yup.



From helmut.kuper at ewetel.de  Tue Mar 23 07:34:02 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 23 Mar 2010 15:34:02 +0100
Subject: [Freeswitch-users] Question about TOS/DSCP
In-Reply-To: <648899B0-74DB-4340-85E8-3FD4B0342477@freeswitch.org>
References: <4BA87DC3.2010004@ewetel.de> <4BA8B5AC.30209@ewetel.de>
	<648899B0-74DB-4340-85E8-3FD4B0342477@freeswitch.org>
Message-ID: <4BA8D15A.1060408@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian,

yes, makes sense ... hm ... sofia sip is only for signalling isn't it?
So the should be a similar way to set DSCP values for FS's inernal RTP
transporter.

Any hint where I can start looking for a temporary hack?


On 23.03.2010 15:00, Brian West wrote:
> the iptables way is better for now.  I thought we exposed those in Sofia but thats just for Signaling and not Media.... I would use the iptables way.
> 


regards helmut
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFLqNFZ4tZeNddg3dwRAjWXAJ96gzkNi8cBD0X1yRn6zdzdls3DsgCdELFw
AsEPQnrWtQwKb+uUGtLqvLY=
=cuvu
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From brian at freeswitch.org  Tue Mar 23 07:41:24 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 09:41:24 -0500
Subject: [Freeswitch-users] Question about TOS/DSCP
In-Reply-To: <4BA8D15A.1060408@ewetel.de>
References: <4BA87DC3.2010004@ewetel.de> <4BA8B5AC.30209@ewetel.de>
	<648899B0-74DB-4340-85E8-3FD4B0342477@freeswitch.org>
	<4BA8D15A.1060408@ewetel.de>
Message-ID: <1621B71B-126D-4779-A1F6-61F2990E28D4@freeswitch.org>

APR has to be patched to allow it.   A patch exists already but its not in the apr tree yet and its not for the version we are using.

/b

On Mar 23, 2010, at 9:34 AM, Helmut Kuper wrote:

> 
> Hi Brian,
> 
> yes, makes sense ... hm ... sofia sip is only for signalling isn't it?
> So the should be a similar way to set DSCP values for FS's inernal RTP
> transporter.
> 
> Any hint where I can start looking for a temporary hack?



From anthony.minessale at gmail.com  Tue Mar 23 08:32:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 10:32:34 -0500
Subject: [Freeswitch-users] FreeSwitch & Vegastream TLS interoperability
	problems?
In-Reply-To: <755C8A7E-6B78-47D4-AF80-224CB0EA2395@freeswitch.org>
References: <d361ca7b1003230019w65b54f7cwbd0c158a86442570@mail.gmail.com>
	<755C8A7E-6B78-47D4-AF80-224CB0EA2395@freeswitch.org>
Message-ID: <191c3a031003230832t11069778m254ef87ecfcd12a6@mail.gmail.com>

They are living in the past since the sips thing is all but dead.
There are no clear specs on tls because as usual there were originally 10
ways to do it.


if that contact is always the same you can manually override it with

<param name="force-register-contact" value="sip:80632 at 10.64.1.6:5061
;transport=tls"/>

in the <user> tag in your directory so when they register the substituted
value will be placed into the DB.





On Tue, Mar 23, 2010 at 8:57 AM, Brian West <brian at freeswitch.org> wrote:

> Every endpoint I have tested that works correctly will append the
> transport=tls so we have a clue what transport to use to call them back.
>
> /b
>
> On Mar 23, 2010, at 2:19 AM, Yehavi Bourvine wrote:
>
> I've contacted the manufacturer who claims that according to RFC5630 it
> should send the request via TLS as it is registered as TLS. I have a feeling
> that the erronous behaviour is because they do not append "transport=tls" to
> the contact field, although they claim it is not required.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From yehavi.bourvine at gmail.com  Tue Mar 23 09:16:15 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 23 Mar 2010 18:16:15 +0200
Subject: [Freeswitch-users] FreeSwitch & Vegastream TLS interoperability
	problems?
In-Reply-To: <191c3a031003230832t11069778m254ef87ecfcd12a6@mail.gmail.com>
References: <d361ca7b1003230019w65b54f7cwbd0c158a86442570@mail.gmail.com>
	<755C8A7E-6B78-47D4-AF80-224CB0EA2395@freeswitch.org>
	<191c3a031003230832t11069778m254ef87ecfcd12a6@mail.gmail.com>
Message-ID: <d361ca7b1003230916y24b9f9a0uab0212bb42f45dd9@mail.gmail.com>

I think my previous response has been lost...

I'll use that as a band-aid and ask the manufacturer to fix his software.

                    Thanks, __Yehavi:

 2010/3/23 Anthony Minessale <anthony.minessale at gmail.com>

> They are living in the past since the sips thing is all but dead.
> There are no clear specs on tls because as usual there were originally 10
> ways to do it.
>
>
> if that contact is always the same you can manually override it with
>
> <param name="force-register-contact" value="sip:80632 at 10.64.1.6:5061
> ;transport=tls"/>
>
> in the <user> tag in your directory so when they register the substituted
> value will be placed into the DB.
>
>
>
>
>
>   On Tue, Mar 23, 2010 at 8:57 AM, Brian West <brian at freeswitch.org>wrote:
>
>>   Every endpoint I have tested that works correctly will append the
>> transport=tls so we have a clue what transport to use to call them back.
>>
>> /b
>>
>>  On Mar 23, 2010, at 2:19 AM, Yehavi Bourvine wrote:
>>
>> I've contacted the manufacturer who claims that according to RFC5630 it
>> should send the request via TLS as it is registered as TLS. I have a feeling
>> that the erronous behaviour is because they do not append "transport=tls" to
>> the contact field, although they claim it is not required.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Mar 23 09:26:51 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 12:26:51 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
	<20100323083208.GA12866@jdc.jasonjgw.net>
Message-ID: <DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>

thanks for your answer Jason.
yes the problem is yet there. before
to submit a bug I want to be sure it's a mistake I did.
so when I set inbound-bypass-media to true, the problem disappear.
it's for sure a media problem.
A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp 
timeout
A phone behind a firewall (tested on 6 PCs too) registers well and doesn't 
make rtp timeout.
I stopped all IP rules like iptables etc... without success.
I'd like to understand

Thanks



----- Original Message ----- 
From: "Jason White" <jason at jasonjgw.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 23, 2010 4:32 AM
Subject: Re: [Freeswitch-users] rtp timeout yet


> Madovsky <infos at madovsky.org> wrote:
>> to be sure I'm silly I downgraded from trunk 17001
>> and rtp timeout disappeared.
>
> Try the latest trunk. If the problem still occurs, capture SIP traces and
> submit a bug report, making sure the configurations are identical.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Mar 23 09:39:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 11:39:10 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
	<20100323083208.GA12866@jdc.jasonjgw.net>
	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
Message-ID: <191c3a031003230939p6ed37680n5396133c2bba7d6@mail.gmail.com>

did you get a pcap trace of the file with tshark or tcpdump?
also consider doing these cli command and reproducing


console loglevel debug
sofia profile <profname> siptrace on (substitute <profname> for your profile
name run it on every profile)

also look at sofia status profile <profname>

and see what the output of that says.

you could post it on pastebin if you need help understanding what it means



On Tue, Mar 23, 2010 at 11:26 AM, Madovsky <infos at madovsky.org> wrote:

> thanks for your answer Jason.
> yes the problem is yet there. before
> to submit a bug I want to be sure it's a mistake I did.
> so when I set inbound-bypass-media to true, the problem disappear.
> it's for sure a media problem.
> A phone with a public IP ( I tested on 6 PCs) registers well, but make a
> rtp
> timeout
> A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
> make rtp timeout.
> I stopped all IP rules like iptables etc... without success.
> I'd like to understand
>
> Thanks
>
>
>
> ----- Original Message -----
> From: "Jason White" <jason at jasonjgw.net>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 23, 2010 4:32 AM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
> > Madovsky <infos at madovsky.org> wrote:
> >> to be sure I'm silly I downgraded from trunk 17001
> >> and rtp timeout disappeared.
> >
> > Try the latest trunk. If the problem still occurs, capture SIP traces and
> > submit a bug report, making sure the configurations are identical.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Tue Mar 23 09:39:57 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 23 Mar 2010 17:39:57 +0100
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>	<20100323083208.GA12866@jdc.jasonjgw.net>
	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
Message-ID: <4BA8EEDD.7000701@infosecurity.ch>

I am having a similar problem with SIP UA based on PJSIP.

My main issue relate with sometimes having one-way audio between two peers.
Exactly the same peers sometimes works, sometime does not work.

What i see is that when everything works are two statement in debug log:
[INFO] switch_rtp.c:2005 Auto Changing port from ...
[INFO] switch_rtp.c:2005 Auto Changing port from ...

When things does not work (one-way audio) i just see 1 line:
[INFO] switch_rtp.c:2005 Auto Changing port from ...

For all calls i am using inbound-proxy-media=true .

Is the same issue you see/encounter?

Fabio

On 23/03/10 17.26, Madovsky wrote:
> thanks for your answer Jason.
> yes the problem is yet there. before
> to submit a bug I want to be sure it's a mistake I did.
> so when I set inbound-bypass-media to true, the problem disappear.
> it's for sure a media problem.
> A phone with a public IP ( I tested on 6 PCs) registers well, but make a rtp
> timeout
> A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
> make rtp timeout.
> I stopped all IP rules like iptables etc... without success.
> I'd like to understand
>
> Thanks
>    



From anthony.minessale at gmail.com  Tue Mar 23 09:51:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 11:51:32 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <4BA8EEDD.7000701@infosecurity.ch>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
	<20100323083208.GA12866@jdc.jasonjgw.net>
	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
	<4BA8EEDD.7000701@infosecurity.ch>
Message-ID: <191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>

That message is a symptom of the UA lying in it's sdp.
it says one ip in the sdp but means another.

We start by sending all the media to the ip specified in the SDP because
that is the only IP we know.
Then there is a small window at the beginning of the call where we assume a
packet sent to us where we specified in our SDP may be a packet from a
mis-configured UA, so we will change our destination to that ip:port to
defeat the bug in the client.

in order to engage this fix the client has to send us several packets so we
can prove it's correct to change our target towards them.


On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> I am having a similar problem with SIP UA based on PJSIP.
>
> My main issue relate with sometimes having one-way audio between two peers.
> Exactly the same peers sometimes works, sometime does not work.
>
> What i see is that when everything works are two statement in debug log:
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
> When things does not work (one-way audio) i just see 1 line:
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
> For all calls i am using inbound-proxy-media=true .
>
> Is the same issue you see/encounter?
>
> Fabio
>
> On 23/03/10 17.26, Madovsky wrote:
> > thanks for your answer Jason.
> > yes the problem is yet there. before
> > to submit a bug I want to be sure it's a mistake I did.
> > so when I set inbound-bypass-media to true, the problem disappear.
> > it's for sure a media problem.
> > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
> rtp
> > timeout
> > A phone behind a firewall (tested on 6 PCs too) registers well and
> doesn't
> > make rtp timeout.
> > I stopped all IP rules like iptables etc... without success.
> > I'd like to understand
> >
> > Thanks
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar 23 09:58:04 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 12:58:04 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>

Holly sh....! , I found the problem !
I didn't know that the fact I don't plug a mic on the legA and talk in it
can causes a rtp timeout !!!!
so I have a question : what's happened if in a conversation  the legB talks 
talks and talks
but legA listens and don't say anything longer as the rtp timeout ? :D
I tried it, and indeed, there's an rtp timeout everytime I stop to talk.




----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 23, 2010 12:26 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


> thanks for your answer Jason.
> yes the problem is yet there. before
> to submit a bug I want to be sure it's a mistake I did.
> so when I set inbound-bypass-media to true, the problem disappear.
> it's for sure a media problem.
> A phone with a public IP ( I tested on 6 PCs) registers well, but make a 
> rtp timeout
> A phone behind a firewall (tested on 6 PCs too) registers well and doesn't 
> make rtp timeout.
> I stopped all IP rules like iptables etc... without success.
> I'd like to understand
>
> Thanks
>
>
>
> ----- Original Message ----- 
> From: "Jason White" <jason at jasonjgw.net>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 23, 2010 4:32 AM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
>> Madovsky <infos at madovsky.org> wrote:
>>> to be sure I'm silly I downgraded from trunk 17001
>>> and rtp timeout disappeared.
>>
>> Try the latest trunk. If the problem still occurs, capture SIP traces and
>> submit a bug report, making sure the configurations are identical.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 



From infos at madovsky.org  Tue Mar 23 10:00:13 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 13:00:13 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>	<20100323083208.GA12866@jdc.jasonjgw.net><DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
	<4BA8EEDD.7000701@infosecurity.ch>
Message-ID: <204E16AA8D53497A9427396B11AC4556@MOBILEE1705>

no I don't use proxy

----- Original Message ----- 
From: "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 23, 2010 12:39 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


>I am having a similar problem with SIP UA based on PJSIP.
>
> My main issue relate with sometimes having one-way audio between two 
> peers.
> Exactly the same peers sometimes works, sometime does not work.
>
> What i see is that when everything works are two statement in debug log:
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
> When things does not work (one-way audio) i just see 1 line:
> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
> For all calls i am using inbound-proxy-media=true .
>
> Is the same issue you see/encounter?
>
> Fabio
>
> On 23/03/10 17.26, Madovsky wrote:
>> thanks for your answer Jason.
>> yes the problem is yet there. before
>> to submit a bug I want to be sure it's a mistake I did.
>> so when I set inbound-bypass-media to true, the problem disappear.
>> it's for sure a media problem.
>> A phone with a public IP ( I tested on 6 PCs) registers well, but make a 
>> rtp
>> timeout
>> A phone behind a firewall (tested on 6 PCs too) registers well and 
>> doesn't
>> make rtp timeout.
>> I stopped all IP rules like iptables etc... without success.
>> I'd like to understand
>>
>> Thanks
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From infos at madovsky.org  Tue Mar 23 10:07:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 13:07:18 -0400
Subject: [Freeswitch-users] rtp timeout yet
Message-ID: <C189D9570E2D4801A9E046B57FD91964@MOBILEE1705>

And to explain why it was working behind a firewall,
I was using another softphone (x-lite) that maybe sends
fake packets to keep the line on...


----- Original Message ----- 
From: "Madovsky" <infos at madovsky.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 23, 2010 12:58 PM
Subject: Re: [Freeswitch-users] rtp timeout yet


> Holly sh....! , I found the problem !
> I didn't know that the fact I don't plug a mic on the legA and talk in it
> can causes a rtp timeout !!!!
> so I have a question : what's happened if in a conversation  the legB 
> talks talks and talks
> but legA listens and don't say anything longer as the rtp timeout ? :D
> I tried it, and indeed, there's an rtp timeout everytime I stop to talk.
>
>
>
>
> ----- Original Message ----- 
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 23, 2010 12:26 PM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
>> thanks for your answer Jason.
>> yes the problem is yet there. before
>> to submit a bug I want to be sure it's a mistake I did.
>> so when I set inbound-bypass-media to true, the problem disappear.
>> it's for sure a media problem.
>> A phone with a public IP ( I tested on 6 PCs) registers well, but make a 
>> rtp timeout
>> A phone behind a firewall (tested on 6 PCs too) registers well and 
>> doesn't make rtp timeout.
>> I stopped all IP rules like iptables etc... without success.
>> I'd like to understand
>>
>> Thanks
>>
>>
>>
>> ----- Original Message ----- 
>> From: "Jason White" <jason at jasonjgw.net>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, March 23, 2010 4:32 AM
>> Subject: Re: [Freeswitch-users] rtp timeout yet
>>
>>
>>> Madovsky <infos at madovsky.org> wrote:
>>>> to be sure I'm silly I downgraded from trunk 17001
>>>> and rtp timeout disappeared.
>>>
>>> Try the latest trunk. If the problem still occurs, capture SIP traces 
>>> and
>>> submit a bug report, making sure the configurations are identical.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
> 



From anthony.minessale at gmail.com  Tue Mar 23 10:09:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 12:09:16 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>
Message-ID: <191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com>

your client probably has vad enabled which is a feature that stops the rtp
when you are not talking.
one thing you can do is disable or increase the duration of that rtp timeout
on your sip profile settings.




On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

> Holly sh....! , I found the problem !
> I didn't know that the fact I don't plug a mic on the legA and talk in it
> can causes a rtp timeout !!!!
> so I have a question : what's happened if in a conversation  the legB talks
> talks and talks
> but legA listens and don't say anything longer as the rtp timeout ? :D
> I tried it, and indeed, there's an rtp timeout everytime I stop to talk.
>
>
>
>
> ----- Original Message -----
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 23, 2010 12:26 PM
> Subject: Re: [Freeswitch-users] rtp timeout yet
>
>
> > thanks for your answer Jason.
> > yes the problem is yet there. before
> > to submit a bug I want to be sure it's a mistake I did.
> > so when I set inbound-bypass-media to true, the problem disappear.
> > it's for sure a media problem.
> > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
> > rtp timeout
> > A phone behind a firewall (tested on 6 PCs too) registers well and
> doesn't
> > make rtp timeout.
> > I stopped all IP rules like iptables etc... without success.
> > I'd like to understand
> >
> > Thanks
> >
> >
> >
> > ----- Original Message -----
> > From: "Jason White" <jason at jasonjgw.net>
> > To: <freeswitch-users at lists.freeswitch.org>
> > Sent: Tuesday, March 23, 2010 4:32 AM
> > Subject: Re: [Freeswitch-users] rtp timeout yet
> >
> >
> >> Madovsky <infos at madovsky.org> wrote:
> >>> to be sure I'm silly I downgraded from trunk 17001
> >>> and rtp timeout disappeared.
> >>
> >> Try the latest trunk. If the problem still occurs, capture SIP traces
> and
> >> submit a bug report, making sure the configurations are identical.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Tue Mar 23 10:27:18 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 13:27:18 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>
	<191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com>
Message-ID: <BF0068ECF0DB46B98E0F41E8453A916B@MOBILEE1705>

I will check in the SIP client libs I use if there is any vad.
od for rtp timeout on profile, but it was useful to be short in case of originator hangs up and
doesn't let the legB on line for long time.
Also what is the role of rtp-hold-timeout-sec ? I tried to change it but has no effect


  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 23, 2010 1:09 PM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  your client probably has vad enabled which is a feature that stops the rtp when you are not talking.
  one thing you can do is disable or increase the duration of that rtp timeout on your sip profile settings.





  On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

    Holly sh....! , I found the problem !
    I didn't know that the fact I don't plug a mic on the legA and talk in it
    can causes a rtp timeout !!!!
    so I have a question : what's happened if in a conversation  the legB talks
    talks and talks
    but legA listens and don't say anything longer as the rtp timeout ? :D
    I tried it, and indeed, there's an rtp timeout everytime I stop to talk.





    ----- Original Message -----
    From: "Madovsky" <infos at madovsky.org>

    To: <freeswitch-users at lists.freeswitch.org>

    Sent: Tuesday, March 23, 2010 12:26 PM
    Subject: Re: [Freeswitch-users] rtp timeout yet



    > thanks for your answer Jason.
    > yes the problem is yet there. before
    > to submit a bug I want to be sure it's a mistake I did.
    > so when I set inbound-bypass-media to true, the problem disappear.
    > it's for sure a media problem.
    > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
    > rtp timeout
    > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
    > make rtp timeout.
    > I stopped all IP rules like iptables etc... without success.
    > I'd like to understand
    >
    > Thanks
    >
    >
    >

    > ----- Original Message -----
    > From: "Jason White" <jason at jasonjgw.net>
    > To: <freeswitch-users at lists.freeswitch.org>
    > Sent: Tuesday, March 23, 2010 4:32 AM
    > Subject: Re: [Freeswitch-users] rtp timeout yet
    >
    >
    >> Madovsky <infos at madovsky.org> wrote:
    >>> to be sure I'm silly I downgraded from trunk 17001
    >>> and rtp timeout disappeared.
    >>
    >> Try the latest trunk. If the problem still occurs, capture SIP traces and
    >> submit a bug report, making sure the configurations are identical.
    >>
    >>
    >> _______________________________________________

    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From infos at madovsky.org  Tue Mar 23 10:33:22 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 13:33:22 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>
	<191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com>
Message-ID: <0B644D3ABC7D4B8E8F64BAC8B113426F@MOBILEE1705>

No it doesnt' seem to use vad, unless you can call vad to another standard name....
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 23, 2010 1:09 PM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  your client probably has vad enabled which is a feature that stops the rtp when you are not talking.
  one thing you can do is disable or increase the duration of that rtp timeout on your sip profile settings.





  On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

    Holly sh....! , I found the problem !
    I didn't know that the fact I don't plug a mic on the legA and talk in it
    can causes a rtp timeout !!!!
    so I have a question : what's happened if in a conversation  the legB talks
    talks and talks
    but legA listens and don't say anything longer as the rtp timeout ? :D
    I tried it, and indeed, there's an rtp timeout everytime I stop to talk.





    ----- Original Message -----
    From: "Madovsky" <infos at madovsky.org>

    To: <freeswitch-users at lists.freeswitch.org>

    Sent: Tuesday, March 23, 2010 12:26 PM
    Subject: Re: [Freeswitch-users] rtp timeout yet



    > thanks for your answer Jason.
    > yes the problem is yet there. before
    > to submit a bug I want to be sure it's a mistake I did.
    > so when I set inbound-bypass-media to true, the problem disappear.
    > it's for sure a media problem.
    > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
    > rtp timeout
    > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
    > make rtp timeout.
    > I stopped all IP rules like iptables etc... without success.
    > I'd like to understand
    >
    > Thanks
    >
    >
    >

    > ----- Original Message -----
    > From: "Jason White" <jason at jasonjgw.net>
    > To: <freeswitch-users at lists.freeswitch.org>
    > Sent: Tuesday, March 23, 2010 4:32 AM
    > Subject: Re: [Freeswitch-users] rtp timeout yet
    >
    >
    >> Madovsky <infos at madovsky.org> wrote:
    >>> to be sure I'm silly I downgraded from trunk 17001
    >>> and rtp timeout disappeared.
    >>
    >> Try the latest trunk. If the problem still occurs, capture SIP traces and
    >> submit a bug report, making sure the configurations are identical.
    >>
    >>
    >> _______________________________________________

    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >> http://www.freeswitch.org
    >


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org




  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From lloyd.aloysius at sunteltech.ca  Tue Mar 23 10:50:31 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Tue, 23 Mar 2010 13:50:31 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
Message-ID: <8a19bf2e1003231050j377d00e1i7a3cfec9cba32d7c@mail.gmail.com>

Hi All,

How to build a dial plan for paging for groups of AASTRA phones. Here is my
environment.

FreeSWITCH on a Public IP address. All the AASTRA phones behind the Linksys
WRT 54GL(Tomato firmware) .

I see the following sample dial plan come with FreeSWITCH.

    <extension name="rtp_multicast_page">
      <condition field="destination_number" expression="^*pagegroup*
$|^7243$">
<action application="answer"/>
<action application="esf_page_group"/>
      </condition>
    </extension>

What is the *pagegroup *variable referring to?


Thanks
Lloyd
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From anthony.minessale at gmail.com  Tue Mar 23 10:52:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 12:52:36 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <0B644D3ABC7D4B8E8F64BAC8B113426F@MOBILEE1705>
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705>
	<191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com>
	<0B644D3ABC7D4B8E8F64BAC8B113426F@MOBILEE1705>
Message-ID: <191c3a031003231052l58ee58e8k68b14bf4bc66ce92@mail.gmail.com>

vad means voice activity detection meaning when you do not speak it does not
send rtp
you can easily test that by running tshark on your shell and see if the
packets stop flowing when you don't talk.

rtp-timeout-sec is how many sec to tolerate no rtp (it should be a very high
value because it's not madatory to send rtp and session timers is a better
way to make sure a call is still up)

rtp-hold-timeout-sec is an alternate most likely even higher value to use
while the call is in a held state assuming you know there will be no audio
during this time it's more acceptable to have a higher number than the
regular timeout.



On Tue, Mar 23, 2010 at 12:33 PM, Madovsky <infos at madovsky.org> wrote:

>  No it doesnt' seem to use vad, unless you can call vad to another
> standard name....
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, March 23, 2010 1:09 PM
> *Subject:* Re: [Freeswitch-users] rtp timeout yet
>
> your client probably has vad enabled which is a feature that stops the rtp
> when you are not talking.
> one thing you can do is disable or increase the duration of that rtp
> timeout on your sip profile settings.
>
>
>
>
> On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:
>
>> Holly sh....! , I found the problem !
>> I didn't know that the fact I don't plug a mic on the legA and talk in it
>> can causes a rtp timeout !!!!
>> so I have a question : what's happened if in a conversation  the legB
>> talks
>> talks and talks
>> but legA listens and don't say anything longer as the rtp timeout ? :D
>> I tried it, and indeed, there's an rtp timeout everytime I stop to talk.
>>
>>
>>
>>
>> ----- Original Message -----
>> From: "Madovsky" <infos at madovsky.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, March 23, 2010 12:26 PM
>> Subject: Re: [Freeswitch-users] rtp timeout yet
>>
>>
>> > thanks for your answer Jason.
>> > yes the problem is yet there. before
>> > to submit a bug I want to be sure it's a mistake I did.
>> > so when I set inbound-bypass-media to true, the problem disappear.
>> > it's for sure a media problem.
>> > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
>> > rtp timeout
>> > A phone behind a firewall (tested on 6 PCs too) registers well and
>> doesn't
>> > make rtp timeout.
>> > I stopped all IP rules like iptables etc... without success.
>> > I'd like to understand
>> >
>> > Thanks
>> >
>> >
>> >
>> > ----- Original Message -----
>> > From: "Jason White" <jason at jasonjgw.net>
>> > To: <freeswitch-users at lists.freeswitch.org>
>> > Sent: Tuesday, March 23, 2010 4:32 AM
>> > Subject: Re: [Freeswitch-users] rtp timeout yet
>> >
>> >
>> >> Madovsky <infos at madovsky.org> wrote:
>> >>> to be sure I'm silly I downgraded from trunk 17001
>> >>> and rtp timeout disappeared.
>> >>
>> >> Try the latest trunk. If the problem still occurs, capture SIP traces
>> and
>> >> submit a bug report, making sure the configurations are identical.
>> >>
>> >>
>> >> _______________________________________________
>>  >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Tue Mar 23 11:00:58 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 23 Mar 2010 19:00:58 +0100
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>	<20100323083208.GA12866@jdc.jasonjgw.net>	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>	<4BA8EEDD.7000701@infosecurity.ch>
	<191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>
Message-ID: <4BA901DA.6050500@infosecurity.ch>

Hi,

now i start to better understand.

I am not using STUN or ICE and in my case clients are in 2 situations:
a) The client has a private ip address between a NAT Port address 
transalation
b) The client has a public ip address but that address is firewalled (3G 
operator that provide a public IP address but does not let incoming 
connection to go trough).

In that scenario i got one-way audio with only one "Auto changing port 
from ..." messages from FS debug log.

Is there a way to force the FS not to look at the IP inside the SDP 
header provisioned by the SIP UA, considering the IP address always 
"buggy", so always not providing the right IP address configuration ?

Fabio

On 23/03/10 17.51, Anthony Minessale wrote:
> That message is a symptom of the UA lying in it's sdp.
> it says one ip in the sdp but means another.
>
> We start by sending all the media to the ip specified in the SDP 
> because that is the only IP we know.
> Then there is a small window at the beginning of the call where we 
> assume a packet sent to us where we specified in our SDP may be a 
> packet from a mis-configured UA, so we will change our destination to 
> that ip:port to defeat the bug in the client.
>
> in order to engage this fix the client has to send us several packets 
> so we can prove it's correct to change our target towards them.
>
>
> On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) 
> <lists at infosecurity.ch <mailto:lists at infosecurity.ch>> wrote:
>
>     I am having a similar problem with SIP UA based on PJSIP.
>
>     My main issue relate with sometimes having one-way audio between
>     two peers.
>     Exactly the same peers sometimes works, sometime does not work.
>
>     What i see is that when everything works are two statement in
>     debug log:
>     [INFO] switch_rtp.c:2005 Auto Changing port from ...
>     [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
>     When things does not work (one-way audio) i just see 1 line:
>     [INFO] switch_rtp.c:2005 Auto Changing port from ...
>
>     For all calls i am using inbound-proxy-media=true .
>
>     Is the same issue you see/encounter?
>
>     Fabio
>
>     On 23/03/10 17.26, Madovsky wrote:
>     > thanks for your answer Jason.
>     > yes the problem is yet there. before
>     > to submit a bug I want to be sure it's a mistake I did.
>     > so when I set inbound-bypass-media to true, the problem disappear.
>     > it's for sure a media problem.
>     > A phone with a public IP ( I tested on 6 PCs) registers well,
>     but make a rtp
>     > timeout
>     > A phone behind a firewall (tested on 6 PCs too) registers well
>     and doesn't
>     > make rtp timeout.
>     > I stopped all IP rules like iptables etc... without success.
>     > I'd like to understand
>     >
>     > Thanks
>     >
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From lloyd.aloysius at gmail.com  Tue Mar 23 11:05:50 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 23 Mar 2010 14:05:50 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
Message-ID: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>

Hi All,

How to build a dial plan for paging for groups of AASTRA phones. Here is my
environment setup.

FreeSWITCH on a Public IP address. All the AASTRA phones behind the Linksys
WRT 54GL(Tomato firmware) .

I see the following sample dialplan come with FreeSWITCH default.xml

    <extension name="rtp_multicast_page">
      <condition field="destination_number" expression="^*pagegroup*
$|^7243$">
<action application="answer"/>
<action application="esf_page_group"/>
      </condition>
    </extension>

What is the *pagegroup *variable referring to?


Thanks
Lloyd
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From infos at madovsky.org  Tue Mar 23 11:07:09 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 14:07:09 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705><191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com><0B644D3ABC7D4B8E8F64BAC8B113426F@MOBILEE1705>
	<191c3a031003231052l58ee58e8k68b14bf4bc66ce92@mail.gmail.com>
Message-ID: <2D3715753CED4E0E92072F7E832DCFD6@MOBILEE1705>

ok thanks I understand now
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 23, 2010 1:52 PM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  vad means voice activity detection meaning when you do not speak it does not send rtp
  you can easily test that by running tshark on your shell and see if the packets stop flowing when you don't talk.

  rtp-timeout-sec is how many sec to tolerate no rtp (it should be a very high value because it's not madatory to send rtp and session timers is a better way to make sure a call is still up)

  rtp-hold-timeout-sec is an alternate most likely even higher value to use while the call is in a held state assuming you know there will be no audio during this time it's more acceptable to have a higher number than the regular timeout.




  On Tue, Mar 23, 2010 at 12:33 PM, Madovsky <infos at madovsky.org> wrote:

    No it doesnt' seem to use vad, unless you can call vad to another standard name....
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 23, 2010 1:09 PM
      Subject: Re: [Freeswitch-users] rtp timeout yet


      your client probably has vad enabled which is a feature that stops the rtp when you are not talking.
      one thing you can do is disable or increase the duration of that rtp timeout on your sip profile settings.





      On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

        Holly sh....! , I found the problem !
        I didn't know that the fact I don't plug a mic on the legA and talk in it
        can causes a rtp timeout !!!!
        so I have a question : what's happened if in a conversation  the legB talks
        talks and talks
        but legA listens and don't say anything longer as the rtp timeout ? :D
        I tried it, and indeed, there's an rtp timeout everytime I stop to talk.





        ----- Original Message -----
        From: "Madovsky" <infos at madovsky.org>

        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Tuesday, March 23, 2010 12:26 PM
        Subject: Re: [Freeswitch-users] rtp timeout yet



        > thanks for your answer Jason.
        > yes the problem is yet there. before
        > to submit a bug I want to be sure it's a mistake I did.
        > so when I set inbound-bypass-media to true, the problem disappear.
        > it's for sure a media problem.
        > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
        > rtp timeout
        > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
        > make rtp timeout.
        > I stopped all IP rules like iptables etc... without success.
        > I'd like to understand
        >
        > Thanks
        >
        >
        >

        > ----- Original Message -----
        > From: "Jason White" <jason at jasonjgw.net>
        > To: <freeswitch-users at lists.freeswitch.org>
        > Sent: Tuesday, March 23, 2010 4:32 AM
        > Subject: Re: [Freeswitch-users] rtp timeout yet
        >
        >
        >> Madovsky <infos at madovsky.org> wrote:
        >>> to be sure I'm silly I downgraded from trunk 17001
        >>> and rtp timeout disappeared.
        >>
        >> Try the latest trunk. If the problem still occurs, capture SIP traces and
        >> submit a bug report, making sure the configurations are identical.
        >>
        >>
        >> _______________________________________________

        >> FreeSWITCH-users mailing list
        >> FreeSWITCH-users at lists.freeswitch.org
        >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >> http://www.freeswitch.org
        >


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org




      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Tue Mar 23 11:09:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 13:09:17 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <4BA901DA.6050500@infosecurity.ch>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
	<20100323083208.GA12866@jdc.jasonjgw.net>
	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
	<4BA8EEDD.7000701@infosecurity.ch>
	<191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>
	<4BA901DA.6050500@infosecurity.ch>
Message-ID: <191c3a031003231109h2f04ad4emfab8922eeda99ef5@mail.gmail.com>

No we don't have such a feature because the media and signaling are
separated by design of SIP.
It would be a horrible assumption to always think the media was coming from
the same IP as the SIP messages.
And even then you still need the port #


On Tue, Mar 23, 2010 at 1:00 PM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

>  Hi,
>
> now i start to better understand.
>
> I am not using STUN or ICE and in my case clients are in 2 situations:
> a) The client has a private ip address between a NAT Port address
> transalation
> b) The client has a public ip address but that address is firewalled (3G
> operator that provide a public IP address but does not let incoming
> connection to go trough).
>
> In that scenario i got one-way audio with only one "Auto changing port from
> ..." messages from FS debug log.
>
> Is there a way to force the FS not to look at the IP inside the SDP header
> provisioned by the SIP UA, considering the IP address always "buggy", so
> always not providing the right IP address configuration ?
>
> Fabio
>
>
> On 23/03/10 17.51, Anthony Minessale wrote:
>
> That message is a symptom of the UA lying in it's sdp.
> it says one ip in the sdp but means another.
>
> We start by sending all the media to the ip specified in the SDP because
> that is the only IP we know.
> Then there is a small window at the beginning of the call where we assume a
> packet sent to us where we specified in our SDP may be a packet from a
> mis-configured UA, so we will change our destination to that ip:port to
> defeat the bug in the client.
>
> in order to engage this fix the client has to send us several packets so we
> can prove it's correct to change our target towards them.
>
>
> On Tue, Mar 23, 2010 at 11:39 AM, Fabio Pietrosanti (naif) <
> lists at infosecurity.ch> wrote:
>
>> I am having a similar problem with SIP UA based on PJSIP.
>>
>> My main issue relate with sometimes having one-way audio between two
>> peers.
>> Exactly the same peers sometimes works, sometime does not work.
>>
>> What i see is that when everything works are two statement in debug log:
>> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>>
>> When things does not work (one-way audio) i just see 1 line:
>> [INFO] switch_rtp.c:2005 Auto Changing port from ...
>>
>> For all calls i am using inbound-proxy-media=true .
>>
>> Is the same issue you see/encounter?
>>
>> Fabio
>>
>> On 23/03/10 17.26, Madovsky wrote:
>> > thanks for your answer Jason.
>> > yes the problem is yet there. before
>> > to submit a bug I want to be sure it's a mistake I did.
>> > so when I set inbound-bypass-media to true, the problem disappear.
>> > it's for sure a media problem.
>> > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
>> rtp
>> > timeout
>> > A phone behind a firewall (tested on 6 PCs too) registers well and
>> doesn't
>> > make rtp timeout.
>> > I stopped all IP rules like iptables etc... without success.
>> > I'd like to understand
>> >
>> > Thanks
>> >
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Tue Mar 23 11:10:23 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 13:10:23 -0500
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
Message-ID: <F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>

Let me say it again... esf_page_group is for snom and linksys multicast paging... I wasn't aware the aastra did this.

/b

On Mar 23, 2010, at 1:05 PM, Aloysius Lloyd wrote:

>     <extension name="rtp_multicast_page">
>       <condition field="destination_number" expression="^pagegroup$|^7243$">
> 	<action application="answer"/>
> 	<action application="esf_page_group"/>
>       </condition>
>     </extension>

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From infos at madovsky.org  Tue Mar 23 11:16:24 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 23 Mar 2010 14:16:24 -0400
Subject: [Freeswitch-users] rtp timeout yet
References: <A45FDCAF30534D4AAB09B9D2FB9EC180@MOBILEE1705><191c3a031003231009g1f59ddeeud51e560aa5c303d8@mail.gmail.com><0B644D3ABC7D4B8E8F64BAC8B113426F@MOBILEE1705>
	<191c3a031003231052l58ee58e8k68b14bf4bc66ce92@mail.gmail.com>
Message-ID: <563D46434EC34BEB98A81774C1E85FBD@MOBILEE1705>

with tshark I get this while I send audio :

121.702216 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 5149 Destination port: netsupport
121.715222 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.735275 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.755342 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.775356 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.795370 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.815371 pu.bl.ic.ip1 -> pu.bl.ic.ip2 UDP Source port: 28922 Destination port: 50644
121.819114 Silicom_0b:00:c3 -> Silicom_0b:00:c3 LLC I, N(R)=0, N(S)=0; DSAP NULL LSAP Individual, SSAP NULL LSAP Command
121.819132 Silicom_0b:00:c3 -> Silicom_0b:00:c3 LLC I, N(R)=0, N(S)=0; DSAP NULL LSAP Individual, SSAP NULL LSAP Command
121.819136 Silicom_0b:00:c3 -> Silicom_0b:00:c3 LLC I, N(R)=0, N(S)=0; DSAP NULL LSAP Individual, SSAP NULL LSAP Command

and indeed, no UDP signal if I stop the audio.



  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, March 23, 2010 1:52 PM
  Subject: Re: [Freeswitch-users] rtp timeout yet


  vad means voice activity detection meaning when you do not speak it does not send rtp
  you can easily test that by running tshark on your shell and see if the packets stop flowing when you don't talk.

  rtp-timeout-sec is how many sec to tolerate no rtp (it should be a very high value because it's not madatory to send rtp and session timers is a better way to make sure a call is still up)

  rtp-hold-timeout-sec is an alternate most likely even higher value to use while the call is in a held state assuming you know there will be no audio during this time it's more acceptable to have a higher number than the regular timeout.




  On Tue, Mar 23, 2010 at 12:33 PM, Madovsky <infos at madovsky.org> wrote:

    No it doesnt' seem to use vad, unless you can call vad to another standard name....
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Tuesday, March 23, 2010 1:09 PM
      Subject: Re: [Freeswitch-users] rtp timeout yet


      your client probably has vad enabled which is a feature that stops the rtp when you are not talking.
      one thing you can do is disable or increase the duration of that rtp timeout on your sip profile settings.





      On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <infos at madovsky.org> wrote:

        Holly sh....! , I found the problem !
        I didn't know that the fact I don't plug a mic on the legA and talk in it
        can causes a rtp timeout !!!!
        so I have a question : what's happened if in a conversation  the legB talks
        talks and talks
        but legA listens and don't say anything longer as the rtp timeout ? :D
        I tried it, and indeed, there's an rtp timeout everytime I stop to talk.





        ----- Original Message -----
        From: "Madovsky" <infos at madovsky.org>

        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Tuesday, March 23, 2010 12:26 PM
        Subject: Re: [Freeswitch-users] rtp timeout yet



        > thanks for your answer Jason.
        > yes the problem is yet there. before
        > to submit a bug I want to be sure it's a mistake I did.
        > so when I set inbound-bypass-media to true, the problem disappear.
        > it's for sure a media problem.
        > A phone with a public IP ( I tested on 6 PCs) registers well, but make a
        > rtp timeout
        > A phone behind a firewall (tested on 6 PCs too) registers well and doesn't
        > make rtp timeout.
        > I stopped all IP rules like iptables etc... without success.
        > I'd like to understand
        >
        > Thanks
        >
        >
        >

        > ----- Original Message -----
        > From: "Jason White" <jason at jasonjgw.net>
        > To: <freeswitch-users at lists.freeswitch.org>
        > Sent: Tuesday, March 23, 2010 4:32 AM
        > Subject: Re: [Freeswitch-users] rtp timeout yet
        >
        >
        >> Madovsky <infos at madovsky.org> wrote:
        >>> to be sure I'm silly I downgraded from trunk 17001
        >>> and rtp timeout disappeared.
        >>
        >> Try the latest trunk. If the problem still occurs, capture SIP traces and
        >> submit a bug report, making sure the configurations are identical.
        >>
        >>
        >> _______________________________________________

        >> FreeSWITCH-users mailing list
        >> FreeSWITCH-users at lists.freeswitch.org
        >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >> http://www.freeswitch.org
        >


        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org




      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_minessale at hotmail.com
      GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:888 at conference.freeswitch.org
      googletalk:conf+888 at conference.freeswitch.org
      pstn:+19193869900



--------------------------------------------------------------------------


      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lloyd.aloysius at gmail.com  Tue Mar 23 11:20:58 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 23 Mar 2010 14:20:58 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
Message-ID: <8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>

Thanks Brian for the verification.

So how to build dailplan for AASTRA phones?

Thanks
Lloyd

On Tue, Mar 23, 2010 at 2:10 PM, Brian West <brian at freeswitch.org> wrote:

> Let me say it again... esf_page_group is for snom and linksys multicast
> paging... I wasn't aware the aastra did this.
>
> /b
>
> On Mar 23, 2010, at 1:05 PM, Aloysius Lloyd wrote:
>
>     <extension name="rtp_multicast_page">
>       <condition field="destination_number" expression="^*pagegroup*
> $|^7243$">
> <action application="answer"/>
> <action application="esf_page_group"/>
>       </condition>
>     </extension>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Mar 23 11:25:13 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 13:25:13 -0500
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
Message-ID: <1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>

see mad boss example in the default dialplan.

/b

On Mar 23, 2010, at 1:20 PM, Aloysius Lloyd wrote:

> Thanks Brian for the verification.
> 
> So how to build dailplan for AASTRA phones?
> 
> Thanks
> Lloyd



From lloyd.aloysius at gmail.com  Tue Mar 23 11:39:17 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 23 Mar 2010 14:39:17 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
	<1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
Message-ID: <8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>

Thanks.

BTW where is the *pagegroup *variable defined for the rtp_multicast_page

Also i find the following from the web

*Multicasting begin to be supported at all major phones manufacturers,
Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast
listening address. This will permit to reduce the generated trafic for an
extensive paging.*



Lloyd


On Tue, Mar 23, 2010 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> see mad boss example in the default dialplan.
>
> /b
>
> On Mar 23, 2010, at 1:20 PM, Aloysius Lloyd wrote:
>
> > Thanks Brian for the verification.
> >
> > So how to build dailplan for AASTRA phones?
> >
> > Thanks
> > Lloyd
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Mar 23 11:43:01 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 13:43:01 -0500
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
	<1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
	<8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>
Message-ID: <36CB8463-DF16-4076-9767-2C7907752A59@freeswitch.org>

It should work then try it out.

/b

On Mar 23, 2010, at 1:39 PM, Aloysius Lloyd wrote:

> Multicasting begin to be supported at all major phones manufacturers, Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast listening address. This will permit to reduce the generated trafic for an extensive paging. 

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From lists at infosecurity.ch  Tue Mar 23 12:10:33 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 23 Mar 2010 20:10:33 +0100
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <191c3a031003231109h2f04ad4emfab8922eeda99ef5@mail.gmail.com>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>	<20100323083208.GA12866@jdc.jasonjgw.net>	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>	<4BA8EEDD.7000701@infosecurity.ch>	<191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>	<4BA901DA.6050500@infosecurity.ch>
	<191c3a031003231109h2f04ad4emfab8922eeda99ef5@mail.gmail.com>
Message-ID: <4BA91229.8060002@infosecurity.ch>

So which is the right way to "stimulate" FS to detect that condition 
quickly?

To send out RTP packets as early as possible from the SIP UA?

What i am still missing is if FS will always rewrite the SDP header to 
act as a media relay between the caller and the called or not, so 
forcing the clients to only send packets to FS.

Fabio

On 23/03/10 19.09, Anthony Minessale wrote:
> No we don't have such a feature because the media and signaling are 
> separated by design of SIP.
> It would be a horrible assumption to always think the media was coming 
> from the same IP as the SIP messages.
> And even then you still need the port #



From anthony.minessale at gmail.com  Tue Mar 23 12:20:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 14:20:50 -0500
Subject: [Freeswitch-users] rtp timeout yet
In-Reply-To: <4BA91229.8060002@infosecurity.ch>
References: <BF89D404049740658F0B636B508894AC@MOBILEE1705>
	<20100323083208.GA12866@jdc.jasonjgw.net>
	<DA593E8E13F948229A0A13A4E747AA1A@MOBILEE1705>
	<4BA8EEDD.7000701@infosecurity.ch>
	<191c3a031003230951r78681b4buee406e5cd0dd39a6@mail.gmail.com>
	<4BA901DA.6050500@infosecurity.ch>
	<191c3a031003231109h2f04ad4emfab8922eeda99ef5@mail.gmail.com>
	<4BA91229.8060002@infosecurity.ch>
Message-ID: <191c3a031003231220l1edac5e6l673a471cd7371b4e@mail.gmail.com>

yes you have to start sending the rtp to FS when you are given the SDP
FS can only detect the error and correct it if it gets several RTP packets.

the other option is to use stun on the client so it can put the right ip in
the SDP to begin with.



On Tue, Mar 23, 2010 at 2:10 PM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> So which is the right way to "stimulate" FS to detect that condition
> quickly?
>
> To send out RTP packets as early as possible from the SIP UA?
>
> What i am still missing is if FS will always rewrite the SDP header to
> act as a media relay between the caller and the called or not, so
> forcing the clients to only send packets to FS.
>
> Fabio
>
> On 23/03/10 19.09, Anthony Minessale wrote:
> > No we don't have such a feature because the media and signaling are
> > separated by design of SIP.
> > It would be a horrible assumption to always think the media was coming
> > from the same IP as the SIP messages.
> > And even then you still need the port #
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From fraserredmond at gmail.com  Tue Mar 23 12:24:15 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Tue, 23 Mar 2010 19:24:15 +0000
Subject: [Freeswitch-users] Simulate dtmf on a channel
Message-ID: <c32ee1cf1003231224h6bf2eceah97108ab5b0e2f2ef@mail.gmail.com>

>From the CLI/event-socket I want to simulate a dtmf press on an existing
channel to make it perform an action as if the caller had pressed the key
themselves (specifically they're in a javascript dialplan and listening to a
session.streamFile() which is waiting for a dtmf input.)

I've had a play with uuid_send_dtmf but all that seems to do is play the
tone on the channel, even with it's undocumented third parameter for
from|to|both. The tone is heard, but the action that would have been
performed if the caller had pressed doesn't happen.

Is there any way to do this?

I'd really like to keep the call in the javascript dialplan, as breaking
them out could get messy, with the other things that are going on.

Cheers,
Fraser
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From jonas.gauffin at gmail.com  Tue Mar 23 12:24:52 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 23 Mar 2010 20:24:52 +0100
Subject: [Freeswitch-users] Problem with bridge to multiple targets and
	loose race.
Message-ID: <d021d3be1003231224k704069eexe589e2b0117893b3@mail.gmail.com>

Hello

I got the following dial plan: http://pastebin.freeswitch.org/12527

Why do the registered phone loose the race after a second? The other number
hasn't even started to ring yet.

336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE]
switch_channel.c:642 New Channel sofia/internal/u1000028
[336a7d1b-3815-4a10-8503-506765035dfe]
336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE]
switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028
[CS_CONSUME_MEDIA] [LOSE_RACE]

Log: http://pastebin.freeswitch.org/12528

//Jonas
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From chris.chen2004 at gmail.com  Tue Mar 23 12:42:14 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 23 Mar 2010 15:42:14 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <36CB8463-DF16-4076-9767-2C7907752A59@freeswitch.org>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
	<1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
	<8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>
	<36CB8463-DF16-4076-9767-2C7907752A59@freeswitch.org>
Message-ID: <507898381003231242i182dcac2q28ae5167da6f0e62@mail.gmail.com>

Hi Lloyd, in the Aastra phone you may set up to receive the multicast paging
by "Group Paging RTP Settings " under "preferences", just put multicast IP
address and port, then you can do multicast paging, however if I am
understanding correctly it only working in LAN environment, I don't think
you can forward multicast traffic through internet unless you sign up the
multicast services.




On Tue, Mar 23, 2010 at 2:43 PM, Brian West <brian at freeswitch.org> wrote:

> It should work then try it out.
>
> /b
>
> On Mar 23, 2010, at 1:39 PM, Aloysius Lloyd wrote:
>
> *Multicasting begin to be supported at all major phones manufacturers,
> Aastra firmware v2.2, Snom v7, Linksys,... allow the setting of a multicast
> listening address. This will permit to reduce the generated trafic for an
> extensive paging.*
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Mar 23 12:46:13 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 14:46:13 -0500
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <507898381003231242i182dcac2q28ae5167da6f0e62@mail.gmail.com>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
	<1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
	<8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>
	<36CB8463-DF16-4076-9767-2C7907752A59@freeswitch.org>
	<507898381003231242i182dcac2q28ae5167da6f0e62@mail.gmail.com>
Message-ID: <3717EF36-F23B-4097-B502-B746E2ED7DA7@freeswitch.org>

Multicast goes over GRE tunnels :P
/b

On Mar 23, 2010, at 2:42 PM, Chris Chen wrote:

> Hi Lloyd, in the Aastra phone you may set up to receive the multicast paging by "Group Paging RTP Settings " under "preferences", just put multicast IP address and port, then you can do multicast paging, however if I am understanding correctly it only working in LAN environment, I don't think you can forward multicast traffic through internet unless you sign up the multicast services.
> 



From anthony.minessale at gmail.com  Tue Mar 23 12:46:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 14:46:52 -0500
Subject: [Freeswitch-users] Problem with bridge to multiple targets and
	loose race.
In-Reply-To: <d021d3be1003231224k704069eexe589e2b0117893b3@mail.gmail.com>
References: <d021d3be1003231224k704069eexe589e2b0117893b3@mail.gmail.com>
Message-ID: <191c3a031003231246q5e2fdcbmc53a18f147c32a33@mail.gmail.com>

you need {ignore_early_media=true} or the first channel to get early media
will be the winner.


On Tue, Mar 23, 2010 at 2:24 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Hello
>
> I got the following dial plan: http://pastebin.freeswitch.org/12527
>
> Why do the registered phone loose the race after a second? The other number
> hasn't even started to ring yet.
>
> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE]
> switch_channel.c:642 New Channel sofia/internal/u1000028
> [336a7d1b-3815-4a10-8503-506765035dfe]
> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE]
> switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028
> [CS_CONSUME_MEDIA] [LOSE_RACE]
>
> Log: http://pastebin.freeswitch.org/12528
>
> //Jonas
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jonas.gauffin at gmail.com  Tue Mar 23 13:08:49 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Tue, 23 Mar 2010 21:08:49 +0100
Subject: [Freeswitch-users] Problem with bridge to multiple targets and
	loose race.
In-Reply-To: <191c3a031003231246q5e2fdcbmc53a18f147c32a33@mail.gmail.com>
References: <d021d3be1003231224k704069eexe589e2b0117893b3@mail.gmail.com>
	<191c3a031003231246q5e2fdcbmc53a18f147c32a33@mail.gmail.com>
Message-ID: <d021d3be1003231308g36c74bfem2e07874585657e0b@mail.gmail.com>

Ok. Setting it per leg didn't help [ignore_early_media=true], but per
channel {ignore_early_media=true} worked.

Thanks!

On Tue, Mar 23, 2010 at 8:46 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you need {ignore_early_media=true} or the first channel to get early media
> will be the winner.
>
>
> On Tue, Mar 23, 2010 at 2:24 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>
>> Hello
>>
>> I got the following dial plan: http://pastebin.freeswitch.org/12527
>>
>> Why do the registered phone loose the race after a second? The other
>> number hasn't even started to ring yet.
>>
>> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE]
>> switch_channel.c:642 New Channel sofia/internal/u1000028
>> [336a7d1b-3815-4a10-8503-506765035dfe]
>> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE]
>> switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028
>> [CS_CONSUME_MEDIA] [LOSE_RACE]
>>
>> Log: http://pastebin.freeswitch.org/12528
>>
>> //Jonas
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Mar 23 13:16:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 15:16:26 -0500
Subject: [Freeswitch-users] Problem with bridge to multiple targets and
	loose race.
In-Reply-To: <d021d3be1003231308g36c74bfem2e07874585657e0b@mail.gmail.com>
References: <d021d3be1003231224k704069eexe589e2b0117893b3@mail.gmail.com>
	<191c3a031003231246q5e2fdcbmc53a18f147c32a33@mail.gmail.com>
	<d021d3be1003231308g36c74bfem2e07874585657e0b@mail.gmail.com>
Message-ID: <191c3a031003231316k769fcc39n5e15b37eb62fe010@mail.gmail.com>

its invalid on a per leg basis, it's only usable globally.


On Tue, Mar 23, 2010 at 3:08 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:

> Ok. Setting it per leg didn't help [ignore_early_media=true], but per
> channel {ignore_early_media=true} worked.
>
> Thanks!
>
> On Tue, Mar 23, 2010 at 8:46 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you need {ignore_early_media=true} or the first channel to get early media
>> will be the winner.
>>
>>
>> On Tue, Mar 23, 2010 at 2:24 PM, Jonas Gauffin <jonas.gauffin at gmail.com>wrote:
>>
>>> Hello
>>>
>>> I got the following dial plan: http://pastebin.freeswitch.org/12527
>>>
>>> Why do the registered phone loose the race after a second? The other
>>> number hasn't even started to ring yet.
>>>
>>> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:14.560000 [NOTICE]
>>> switch_channel.c:642 New Channel sofia/internal/u1000028
>>> [336a7d1b-3815-4a10-8503-506765035dfe]
>>> 336a7d1b-3815-4a10-8503-506765035dfe 2010-03-23 20:06:15.300000 [NOTICE]
>>> switch_ivr_originate.c:3060 Hangup sofia/internal/u1000028
>>> [CS_CONSUME_MEDIA] [LOSE_RACE]
>>>
>>> Log: http://pastebin.freeswitch.org/12528
>>>
>>> //Jonas
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lloyd.aloysius at gmail.com  Tue Mar 23 13:17:26 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Tue, 23 Mar 2010 16:17:26 -0400
Subject: [Freeswitch-users] Paging - AASTRA phones
In-Reply-To: <3717EF36-F23B-4097-B502-B746E2ED7DA7@freeswitch.org>
References: <8a19bf2e1003231105r6390c67elfc3cf57e57d01d64@mail.gmail.com>
	<F29CE665-5B72-47D7-8566-46F3A4487E17@freeswitch.org>
	<8a19bf2e1003231120l7b6b253cle6187636da0d4e9f@mail.gmail.com>
	<1D2C1B0B-1E77-4EBC-AB9E-48292B8253D7@freeswitch.org>
	<8a19bf2e1003231139k133f918en7a0f59337b7654db@mail.gmail.com>
	<36CB8463-DF16-4076-9767-2C7907752A59@freeswitch.org>
	<507898381003231242i182dcac2q28ae5167da6f0e62@mail.gmail.com>
	<3717EF36-F23B-4097-B502-B746E2ED7DA7@freeswitch.org>
Message-ID: <8a19bf2e1003231317t214cad50nb0c95207f05717f1@mail.gmail.com>

Chris & Brian, Thanks.

I finally manged the paging working for AASTRA phones. This is modified
intercom dialplan with group_call


<extension name="extension-paging">
      <condition field="destination_number" expression="^8000$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="sip_auto_answer=true"/>
<action application="bridge" data="${group_call(pagegroup@
${domain_name})}"/>
      </condition>
    </extension>

I will try the multicast with a  Cisco Router support GRE

Lloyd
On Tue, Mar 23, 2010 at 3:46 PM, Brian West <brian at freeswitch.org> wrote:

> Multicast goes over GRE tunnels :P
> /b
>
> On Mar 23, 2010, at 2:42 PM, Chris Chen wrote:
>
> > Hi Lloyd, in the Aastra phone you may set up to receive the multicast
> paging by "Group Paging RTP Settings " under "preferences", just put
> multicast IP address and port, then you can do multicast paging, however if
> I am understanding correctly it only working in LAN environment, I don't
> think you can forward multicast traffic through internet unless you sign up
> the multicast services.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juancarlosg6 at gmail.com  Tue Mar 23 13:14:42 2010
From: juancarlosg6 at gmail.com (Juan Carlos Gonzalez Cardona)
Date: Tue, 23 Mar 2010 15:14:42 -0500
Subject: [Freeswitch-users] Time limit
Message-ID: <d71129941003231314j66e2a0b5kc380c38c7c1c7c4b@mail.gmail.com>

Hi everyone, is posible use call limite for calls, to some extensiones?

In Asterisk is possible creating custom context with set time limit to
specific trunk, but i see that Freeswitch is more stable code, i need to
know if is possible create groups for limit calls and assign extensiones to
that group, if is possible, how i do that, thank?s.

-- 
Juan Carlos Gonzalez C.
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From jcasale at activenetwerx.com  Tue Mar 23 14:01:20 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 23 Mar 2010 21:01:20 +0000
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
	<CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
	<486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E5F76@prato.activenetwerx.local>

>What is the output of the build when you type make?

Ok, I didn't have access to that machine today, but on another
following my notes from previous builds, I got the same errors
again.

I presume I missing a dep (CentOS 5 x64) but I compared my notes
with the wiki and they match.

make (both stdout/stderr) are pastebin'ed:
http://pastebin.freeswitch.org/12529

Thanks!
jlc


From rupa at rupa.com  Tue Mar 23 14:03:12 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 23 Mar 2010 16:03:12 -0500
Subject: [Freeswitch-users] Time limit
In-Reply-To: <d71129941003231314j66e2a0b5kc380c38c7c1c7c4b@mail.gmail.com>
References: <d71129941003231314j66e2a0b5kc380c38c7c1c7c4b@mail.gmail.com>
Message-ID: <b39273531003231403s3857db6dv314b4cfd682956a8@mail.gmail.com>

for time limit, look at sched_hangup or sched_transfer
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_transfer

If you want to limit the number of calls, look at mod_limit:
http://wiki.freeswitch.org/wiki/Mod_limit

On Tue, Mar 23, 2010 at 3:14 PM, Juan Carlos Gonzalez Cardona <
juancarlosg6 at gmail.com> wrote:

>
> Hi everyone, is posible use call limite for calls, to some extensiones?
>
> In Asterisk is possible creating custom context with set time limit to
> specific trunk, but i see that Freeswitch is more stable code, i need to
> know if is possible create groups for limit calls and assign extensiones to
> that group, if is possible, how i do that, thank?s.
>
> --
> Juan Carlos Gonzalez C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian at freeswitch.org  Tue Mar 23 14:13:12 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 16:13:12 -0500
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E5F76@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
	<CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
	<486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E98E5F76@prato.activenetwerx.local>
Message-ID: <2EA2C4F7-486E-4457-A7C9-833889FB821A@freeswitch.org>

/usr/bin/ld: cannot find -ljpeg

install libjpeg-devel and you should be set.. its not an OpenZAP compile issue you're simply failing to pay closer attention to the errors.

/b

On Mar 23, 2010, at 4:01 PM, Joseph L. Casale wrote:

>> What is the output of the build when you type make?
> 
> Ok, I didn't have access to that machine today, but on another
> following my notes from previous builds, I got the same errors
> again.
> 
> I presume I missing a dep (CentOS 5 x64) but I compared my notes
> with the wiki and they match.
> 
> make (both stdout/stderr) are pastebin'ed:
> http://pastebin.freeswitch.org/12529
> 
> Thanks!
> jlc
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From christian.loeschenkohl at xpirio.com  Tue Mar 23 14:38:10 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 23 Mar 2010 22:38:10 +0100
Subject: [Freeswitch-users] donating german prompts
Message-ID: <4BA934C2.3050209@xpirio.com>

hello

we are going to produce a full set of german audio prompts and donate them to this wonderful project.
i have attached a prompt list with the phrases we are going to record.

the source of this prompt list is the file docs/phrase/phrase_de.xml. i have also compared it with
the english file under docs/phrase/phrase_en.xml an merged this two + translated what was needed/had
not been present in the german langunge file from english to german.

i would be happy if someone could give me feedback on this and at best review the phrase list in terms
of correctness and completeness.

br


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com
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From javieraristizabal at gmail.com  Tue Mar 23 14:57:16 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Tue, 23 Mar 2010 16:57:16 -0500
Subject: [Freeswitch-users] mod_xml_curl question
Message-ID: <b6f11f301003231457h472ee8fdn1cb9e6000753a64d@mail.gmail.com>

Hi folks. :)

is it possible to serve up you gateway configs but keep your sip profile
stuff static? if yes!,  could  you give me clues, plz?



Many thanks


/Javier
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From devel at thom.fr.eu.org  Tue Mar 23 15:09:34 2010
From: devel at thom.fr.eu.org (devel at thom.fr.eu.org)
Date: Tue, 23 Mar 2010 23:09:34 +0100
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <A643303450404497B14F79175BEC0EF4@MOBILEE1705>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705><191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com><3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705><e2459a682c4993206a5701120a82ed33@thom.fr.eu.org><49B3842E181347948484F22AB2FABBD0@MOBILEE1705><191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com><F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
	<A643303450404497B14F79175BEC0EF4@MOBILEE1705>
Message-ID: <005801cacad5$8a009820$9e01c860$@fr.eu.org>

So I did continue to track that problem.

 

I found that when running sendmail from freeswitch, it seems that the setgid
bit on sendmail is not honored (I get messages like sendmail[13534]:
NOQUEUE: SYSERR(freeswitch): can not

 chdir(/var/spool/mqueue-client/): Permission denied or NOQUEUE:
SYSERR(UID101): can not wri

te to queue directory /var/spool/mqueue-client/ (RunAsGid=102,
required=107): Pe

rmission denied when I gave freeswitch the permission on
/var/spool/mqueue-client).

In any case that ends up to sendmail segfault, maybe due to a stack problem.

 

I could solve this by installing msmtp (it can live aside sendmail without
breaking my messaging setup, at least on debian)

 

Fran?ois

 

De : freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de
Madovsky
Envoy? : samedi 20 mars 2010 18:13
? : freeswitch-users at lists.freeswitch.org
Objet : Re: [Freeswitch-users] voicemail options [temporarly solved]

 

ok irght.

So for now I will continue to use my patch to make it work.

I will give to you the status of any system upgrade that resolves the
problem

----- Original Message ----- 

From: Anthony Minessale <mailto:anthony.minessale at gmail.com>  

To: freeswitch-users at lists.freeswitch.org 

Sent: Saturday, March 20, 2010 12:58 PM

Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]

 

I am willing to change anything we have to as long as it works on all
platforms and we can give proper warnings to others.
Maybe we will do something like another param to say if your mailer app
needs a pipe or can accept the filename so we can do both
and leave the defaults to the current way.

Maybe we can also have another option to make a simple relay code internally
that can deliver the file to a mail relay if you set that option.




On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:

Ok I undertand.

buy Why to continue to use pipe and STDIN since to use "filename" path works
?

----- Original Message ----- 

From: Anthony Minessale <mailto:anthony.minessale at gmail.com>  

To: freeswitch-users at lists.freeswitch.org 

Sent: Saturday, March 20, 2010 12:12 PM

Subject: Re: [Freeswitch-users] voicemail options [temporarly solved]

 

I think the problem is that fs sets the stack size for the core and new
threads to 240k so when the child is born from system() the new process
inherits the small stack size and sendmail needs more than 240k

If you are root, it has the priveledge to raise the stack size to the
default size of 8m and sets it back down when the command completes.

If you are not root you lack these privs and the stack size remains small.

This has been a long time problem how to make a solution that is cross
platform and will not break working configurations. 

On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:

I tried also with sudo and freeswitch user in sudoers with the same result,
segmentation fault.

Anyway my patch works perfectly now, even with sendmail (the only difference
without patch is the "From" is taken from the voicemail.tpl and not the user
variable)


>
> ----- Original Message -----
> From: Fran?ois Legal
> To: freeswitch-users at lists.freeswitch.org


> Sent: Friday, March 19, 2010 6:51 AM
> Subject: Re: [Freeswitch-users] voicemail options [tempora...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

...

  _____  

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

  _____  

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Tue Mar 23 15:29:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Mar 2010 17:29:19 -0500
Subject: [Freeswitch-users] voicemail options [temporarly solved]
In-Reply-To: <005801cacad5$8a009820$9e01c860$@fr.eu.org>
References: <1AAB7BE8678D457A986192F5B233490B@MOBILEE1705>
	<191c3a031003181310ob1f1274u621a68b07fe5f522@mail.gmail.com>
	<3DA0B3F3EA3D434E8C66208B804C5706@MOBILEE1705>
	<e2459a682c4993206a5701120a82ed33@thom.fr.eu.org>
	<49B3842E181347948484F22AB2FABBD0@MOBILEE1705>
	<191c3a031003200912r3e0bd8f9x22a7ec776e3a1619@mail.gmail.com>
	<F135233BD49A4809B1EA5A399BFA7E95@MOBILEE1705>
	<191c3a031003200958t6e31f055g499be1153e7c4433@mail.gmail.com>
	<A643303450404497B14F79175BEC0EF4@MOBILEE1705>
	<005801cacad5$8a009820$9e01c860$@fr.eu.org>
Message-ID: <191c3a031003231529w5e940113y899bcbfeff10d0eb@mail.gmail.com>

lets open a jira about it as a reminder.
we will figure out what to do there and come up with something.


On Tue, Mar 23, 2010 at 5:09 PM, <devel at thom.fr.eu.org> wrote:

>  So I did continue to track that problem.
>
>
>
> I found that when running sendmail from freeswitch, it seems that the
> setgid bit on sendmail is not honored (I get messages like sendmail[13534]:
> NOQUEUE: SYSERR(freeswitch): can not
>
>  chdir(/var/spool/mqueue-client/): Permission denied or NOQUEUE:
> SYSERR(UID101): can not wri
>
> te to queue directory /var/spool/mqueue-client/ (RunAsGid=102,
> required=107): Pe
>
> rmission denied when I gave freeswitch the permission on
> /var/spool/mqueue-client).
>
> In any case that ends up to sendmail segfault, maybe due to a stack
> problem.
>
>
>
> I could solve this by installing msmtp (it can live aside sendmail without
> breaking my messaging setup, at least on debian)
>
>
>
> Fran?ois
>
>
>
> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Madovsky
> *Envoy? :* samedi 20 mars 2010 18:13
> *? :* freeswitch-users at lists.freeswitch.org
> *Objet :* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
>
>
> ok irght.
>
> So for now I will continue to use my patch to make it work.
>
> I will give to you the status of any system upgrade that resolves the
> problem
>
>  ----- Original Message -----
>
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Sent:* Saturday, March 20, 2010 12:58 PM
>
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
>
>
> I am willing to change anything we have to as long as it works on all
> platforms and we can give proper warnings to others.
> Maybe we will do something like another param to say if your mailer app
> needs a pipe or can accept the filename so we can do both
> and leave the defaults to the current way.
>
> Maybe we can also have another option to make a simple relay code
> internally that can deliver the file to a mail relay if you set that option.
>
>
>  On Sat, Mar 20, 2010 at 11:47 AM, Madovsky <infos at madovsky.org> wrote:
>
> Ok I undertand.
>
> buy Why to continue to use pipe and STDIN since to use "filename" path
> works ?
>
>  ----- Original Message -----
>
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Sent:* Saturday, March 20, 2010 12:12 PM
>
> *Subject:* Re: [Freeswitch-users] voicemail options [temporarly solved]
>
>
>
> I think the problem is that fs sets the stack size for the core and new
> threads to 240k so when the child is born from system() the new process
> inherits the small stack size and sendmail needs more than 240k
>
> If you are root, it has the priveledge to raise the stack size to the
> default size of 8m and sets it back down when the command completes.
>
> If you are not root you lack these privs and the stack size remains small.
>
> This has been a long time problem how to make a solution that is cross
> platform and will not break working configurations.
>
> On Mar 19, 2010 6:47 AM, "Madovsky" <infos at madovsky.org> wrote:
>
> I tried also with sudo and freeswitch user in sudoers with the same result,
> segmentation fault.
>
> Anyway my patch works perfectly now, even with sendmail (the only
> difference without patch is the "From" is taken from the voicemail.tpl and
> not the user variable)
>
>
> >
> > ----- Original Message -----
> > From: Fran?ois Legal
> > To: freeswitch-users at lists.freeswitch.org
>
>
> > Sent: Friday, March 19, 2010 6:51 AM
> > Subject: Re: [Freeswitch-users] voicemail options [tempora...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ...
>
>   ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Tue Mar 23 15:47:16 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 23 Mar 2010 22:47:16 +0000
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <2EA2C4F7-486E-4457-A7C9-833889FB821A@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
	<CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
	<486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E98E5F76@prato.activenetwerx.local>
	<2EA2C4F7-486E-4457-A7C9-833889FB821A@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E61E2@prato.activenetwerx.local>

>/usr/bin/ld: cannot find -ljpeg
>
>install libjpeg-devel and you should be set.. its not an OpenZAP compile issue you're simply failing to pay closer attention to the errors.

Something is up with the original server, I need to see what is different
between it and this test box (nothing should be), but thanks for that pointer.

Would I need to rerun ./configure now after installing libjpeg-devel?

Lastly, how does one avoid re-downloading the sound tar's in subsequent
make's?

Thanks,
jlc


From msc at freeswitch.org  Tue Mar 23 15:48:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Mar 2010 15:48:04 -0700
Subject: [Freeswitch-users] donating german prompts
In-Reply-To: <4BA934C2.3050209@xpirio.com>
References: <4BA934C2.3050209@xpirio.com>
Message-ID: <87f2f3b91003231548h7183c96eu4a093f6c441fd042@mail.gmail.com>

Christian,

This is fantastic! Thank you very much for giving back to the community. We
definitely appreciate it.
-MC

2010/3/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> we are going to produce a full set of german audio prompts and donate them
> to this wonderful project.
> i have attached a prompt list with the phrases we are going to record.
>
> the source of this prompt list is the file docs/phrase/phrase_de.xml. i
> have also compared it with
> the english file under docs/phrase/phrase_en.xml an merged this two +
> translated what was needed/had
> not been present in the german langunge file from english to german.
>
> i would be happy if someone could give me feedback on this and at best
> review the phrase list in terms
> of correctness and completeness.
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Mar 23 15:58:45 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 23 Mar 2010 17:58:45 -0500
Subject: [Freeswitch-users] Compile issue with openzap
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E61E2@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E483A@prato.activenetwerx.local>
	<530D86B5-F38D-4E94-AC8F-BF0BF3125D66@freeswitch.org>
	<20100323035619.GA11860@jdc.jasonjgw.net>
	<CA5A491E9DEFBE4CB777DE97E21575E98E4CFF@prato.activenetwerx.local>
	<486883A1-8BB1-4318-82B5-4505D0B53EF3@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E98E5F76@prato.activenetwerx.local>
	<2EA2C4F7-486E-4457-A7C9-833889FB821A@freeswitch.org>
	<CA5A491E9DEFBE4CB777DE97E21575E98E61E2@prato.activenetwerx.local>
Message-ID: <DE1D2CC3-DC02-4FBF-8569-AC2389A0064B@freeswitch.org>

type "make current" if you have SVN installed it will work.

/b

On Mar 23, 2010, at 5:47 PM, Joseph L. Casale wrote:

> Something is up with the original server, I need to see what is different
> between it and this test box (nothing should be), but thanks for that pointer.
> 
> Would I need to rerun ./configure now after installing libjpeg-devel?
> 
> Lastly, how does one avoid re-downloading the sound tar's in subsequent
> make's?
> 
> Thanks,
> jlc



From woodydickson at gmail.com  Tue Mar 23 17:26:13 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 24 Mar 2010 08:26:13 +0800
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
Message-ID: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>

Hello,

Freeswitch's ACD is cool!

I am having a strange problem here.  If I use a dialplan extension to
bring a caller to a FIFO ( that already have the agent added using
"fifo_member add" command), then, the agent's phone will ring and call
is successfully bridged when the agent picks up the phone.

The problem I have is that if the caller is bought to the FIFO after
pressing a specific key in an IVR, then, when the agent picks up the
phone, both caller and agent are being hung up by freeswitch.  In the
console, it says that it is failed to bridge the caller to the agent.

Why is that the case and how can I resolve that?

Best Regards,
Woody


From msc at freeswitch.org  Tue Mar 23 17:27:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Mar 2010 17:27:56 -0700
Subject: [Freeswitch-users] how to export variables from b-leg to a-leg?
In-Reply-To: <50e456911003190601i395d108cg55c98a214e84eeae@mail.gmail.com>
References: <50e456911003190601i395d108cg55c98a214e84eeae@mail.gmail.com>
Message-ID: <87f2f3b91003231727x7740cb0fs16e7338d72138088@mail.gmail.com>

On Fri, Mar 19, 2010 at 6:01 AM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Hi guys,
> Freeswitch Rocks!
>
> Does anybody know how to export variables from the b-leg to a-leg...
> I start a bridge from lua on an outgoing gateway and on the answer of
> the b-leg I would like to set a variable on the a-leg...
>
> <action application="export" data="nolocal:execute_on_answer=lua
> cc.lua extensions_answer ${uuid}"/>
> <bridge ....>
>
> from cc.lua I try:
> session:execute("export", "foo=123");
>
> but no success...
> any hint?
>

Sorry, I meant to discuss this earlier. If you know the uuid of the A leg
then just use uuid_setvar API:

api = freeswitch.API();
res = api:execute("uuid_setvar", a_leg_uuid .. " " .. chan_var .. " " ..
chan_var_val)

-MC
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From msc at freeswitch.org  Tue Mar 23 17:48:54 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Mar 2010 17:48:54 -0700
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
In-Reply-To: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
References: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
Message-ID: <87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>

On Tue, Mar 23, 2010 at 5:26 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hello,
>
> Freeswitch's ACD is cool!
>
> I am having a strange problem here.  If I use a dialplan extension to
> bring a caller to a FIFO ( that already have the agent added using
> "fifo_member add" command), then, the agent's phone will ring and call
> is successfully bridged when the agent picks up the phone.
>
> The problem I have is that if the caller is bought to the FIFO after
> pressing a specific key in an IVR, then, when the agent picks up the
> phone, both caller and agent are being hung up by freeswitch.  In the
> console, it says that it is failed to bridge the caller to the agent.
>
> Why is that the case and how can I resolve that?
>

Snag a debug log of this happening and drop it on pastebin. Usually the
debug log will have the clues you need. Put the pastebin link in this thread
and we'll take a look.
-MC
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From woodydickson at gmail.com  Tue Mar 23 18:46:33 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 24 Mar 2010 09:46:33 +0800
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
In-Reply-To: <87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>
References: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
	<87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>
Message-ID: <a44688a1003231846w4a63b39eqe3cac4f8b608dcca@mail.gmail.com>

Hello,

Here is the log: http://pastebin.freeswitch.org/12533

I pastebin the relevant debug log.  The highlighted line shows:

#
2010-03-24 09:34:00.673468 [WARNING] switch_ivr_bridge.c:1233 Bridge
Failed sofia/internal/sip:200003 at 192.168.1.13:9000->sofia/internal/200002 at 192.168.1.102

Please help.

Best Regards,
Woody

On Wed, Mar 24, 2010 at 8:48 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, Mar 23, 2010 at 5:26 PM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>>
>> Hello,
>>
>> Freeswitch's ACD is cool!
>>
>> I am having a strange problem here. ?If I use a dialplan extension to
>> bring a caller to a FIFO ( that already have the agent added using
>> "fifo_member add" command), then, the agent's phone will ring and call
>> is successfully bridged when the agent picks up the phone.
>>
>> The problem I have is that if the caller is bought to the FIFO after
>> pressing a specific key in an IVR, then, when the agent picks up the
>> phone, both caller and agent are being hung up by freeswitch. ?In the
>> console, it says that it is failed to bridge the caller to the agent.
>>
>> Why is that the case and how can I resolve that?
>
> Snag a debug log of this happening and drop it on pastebin. Usually the
> debug log will have the clues you need. Put the pastebin link in this thread
> and we'll take a look.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jcasale at activenetwerx.com  Tue Mar 23 21:37:38 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 04:37:38 +0000
Subject: [Freeswitch-users] Success with a TDM410P finally
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>

Finally got this working but have a couple simple things to work out.

The gain seems rather excessive, I assumed zt.conf's rx|txgain was
where this needed to be adjusted, but values did not have an effect
there.

I have disabled the echo canceller in zt.conf as per the recent thread
and the results with the mg2 canceller are not very good, is it also
required to be disabled when using any echo canceller in dahdi?


Thanks,
jlc


From nandy1925 at gmail.com  Tue Mar 23 22:03:02 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 24 Mar 2010 13:03:02 +0800
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
Message-ID: <7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>

hi joseph,

have u tested the Oslec echo-canceller?

-nandy

On Wed, Mar 24, 2010 at 12:37 PM, Joseph L. Casale <
jcasale at activenetwerx.com> wrote:

> Finally got this working but have a couple simple things to work out.
>
> The gain seems rather excessive, I assumed zt.conf's rx|txgain was
> where this needed to be adjusted, but values did not have an effect
> there.
>
> I have disabled the echo canceller in zt.conf as per the recent thread
> and the results with the mg2 canceller are not very good, is it also
> required to be disabled when using any echo canceller in dahdi?
>
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Mar 23 22:16:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Mar 2010 22:16:14 -0700
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
In-Reply-To: <a44688a1003231846w4a63b39eqe3cac4f8b608dcca@mail.gmail.com>
References: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
	<87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>
	<a44688a1003231846w4a63b39eqe3cac4f8b608dcca@mail.gmail.com>
Message-ID: <87f2f3b91003232216t64253560uf49984a94d485ee@mail.gmail.com>

On Tue, Mar 23, 2010 at 6:46 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hello,
>
> Here is the log: http://pastebin.freeswitch.org/12533
>
> I pastebin the relevant debug log.  The highlighted line shows:
>
> #
> 2010-03-24 09:34:00.673468 [WARNING] switch_ivr_bridge.c:1233 Bridge
> Failed sofia/internal/sip:200003 at 192.168.1.13:9000->sofia/internal/
> 200002 at 192.168.1.102
>
> Please help.
>
> Best Regards,
> Woody
>
> Yeah, the debug log didn't really say why the bridge failed. I'd try a few
things: pastebin your relevant dialplan bits as well as whatever code you
use to put the agents into the queue as members. Also, turn on SIP debugging
at the console:
sofia profile internal siptrace on

Look to see if there's anything interesting or unusual in the SIP dialog.
You can pastebin that as well.
-MC
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From infos at madovsky.org  Tue Mar 23 22:40:26 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Mar 2010 01:40:26 -0400
Subject: [Freeswitch-users] leg B hangup
Message-ID: <A01715D0E49F441D8CBCC334AC9DDC6E@MOBILEE1705>

Is there any variable  to set to help leg B hangup when
Leg A (originator call) hangs up or is it a a normal behaviour ?

Thanks

Fanck
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From jonas.gauffin at gmail.com  Tue Mar 23 22:51:55 2010
From: jonas.gauffin at gmail.com (Jonas Gauffin)
Date: Wed, 24 Mar 2010 06:51:55 +0100
Subject: [Freeswitch-users] how to export variables from b-leg to a-leg?
In-Reply-To: <87f2f3b91003231727x7740cb0fs16e7338d72138088@mail.gmail.com>
References: <50e456911003190601i395d108cg55c98a214e84eeae@mail.gmail.com>
	<87f2f3b91003231727x7740cb0fs16e7338d72138088@mail.gmail.com>
Message-ID: <d021d3be1003232251j437a5dc5o7e69a4a3e99f8e72@mail.gmail.com>

Can it be done in the dial plan, and without knowing the UUID?

I'm using the regular bind_meta_app and transfer in the feature dialplan. I
need to set a special variable for my system when the actual transfer
command is about to be executed in the dial plan on the a-leg.

On Wed, Mar 24, 2010 at 1:27 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Mar 19, 2010 at 6:01 AM, rentmycoder rentmycoder <
> rentmycoder at gmail.com> wrote:
>
>> Hi guys,
>> Freeswitch Rocks!
>>
>> Does anybody know how to export variables from the b-leg to a-leg...
>> I start a bridge from lua on an outgoing gateway and on the answer of
>> the b-leg I would like to set a variable on the a-leg...
>>
>> <action application="export" data="nolocal:execute_on_answer=lua
>> cc.lua extensions_answer ${uuid}"/>
>> <bridge ....>
>>
>> from cc.lua I try:
>> session:execute("export", "foo=123");
>>
>> but no success...
>> any hint?
>>
>
> Sorry, I meant to discuss this earlier. If you know the uuid of the A leg
> then just use uuid_setvar API:
>
> api = freeswitch.API();
> res = api:execute("uuid_setvar", a_leg_uuid .. " " .. chan_var .. " " ..
> chan_var_val)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Wed Mar 24 00:24:52 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 24 Mar 2010 15:24:52 +0800
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
In-Reply-To: <87f2f3b91003232216t64253560uf49984a94d485ee@mail.gmail.com>
References: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
	<87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>
	<a44688a1003231846w4a63b39eqe3cac4f8b608dcca@mail.gmail.com>
	<87f2f3b91003232216t64253560uf49984a94d485ee@mail.gmail.com>
Message-ID: <a44688a1003240024i1f5b4dc9jc74f7d3f570b0851@mail.gmail.com>

Hi Michael,

Thanks in advance for all your help.

I pastebin the SIP log as well at http://pastebin.freeswitch.org/

It does not seem to tell much.

Best Regards,
Woody

On Wed, Mar 24, 2010 at 1:16 PM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, Mar 23, 2010 at 6:46 PM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>>
>> Hello,
>>
>> Here is the log: http://pastebin.freeswitch.org/12533
>>
>> I pastebin the relevant debug log. ?The highlighted line shows:
>>
>> #
>> 2010-03-24 09:34:00.673468 [WARNING] switch_ivr_bridge.c:1233 Bridge
>> Failed
>> sofia/internal/sip:200003 at 192.168.1.13:9000->sofia/internal/200002 at 192.168.1.102
>>
>> Please help.
>>
>> Best Regards,
>> Woody
>>
> Yeah, the debug log didn't really say why the bridge failed. I'd try a few
> things: pastebin your relevant dialplan bits as well as whatever code you
> use to put the agents into the queue as members. Also, turn on SIP debugging
> at the console:
> sofia profile internal siptrace on
>
> Look to see if there's anything interesting or unusual in the SIP dialog.
> You can pastebin that as well.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jcasale at activenetwerx.com  Wed Mar 24 03:31:57 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 10:31:57 +0000
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
	<7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>

>hi joseph,
>
>have u tested the Oslec echo-canceller?

Nandy,
I'd love to, but I have never had any luck getting it to work.
Currently, I am using dahdi packages from digiums yum repo. Are
you using rpm's or compiling from source?

Thanks,
jlc 


From jcasale at activenetwerx.com  Wed Mar 24 03:53:08 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 10:53:08 +0000
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
	<7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E6C36@prato.activenetwerx.local>

>hi joseph,
>
>have u tested the Oslec echo-canceller?

Looking back through my notes I see Anthony Messina suggested I use
his build script to create the rpms w/ oslec enabled which I just did
except that the script at http://messinet.com/trac/rpms/browser/tools/svn-build-rpm
doesn?t build dahdi-tools-libs which is required.

Anyone know the workaround?

Thanks guys,
jlc


From Russell.Mosemann at cune.org  Wed Mar 24 04:16:13 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 24 Mar 2010 06:16:13 -0500
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local><7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>
Message-ID: <7790F777BD4A41B28FCC6DC1A4A7CE2F@cune.pri>

Joseph L. Casale wrote:

> Currently, I am using dahdi packages from digiums yum repo. Are
> you using rpm's or compiling from source?

We're using OSLEC with DAHDI compiled from source. Follow the directions on the OSLEC web page for DAHDI. We haven't tested it extensively, because our system is just for playing around, but it seems to be working fine. You will, of course, get the best results from a good card with hardware echo cancellation.

--
Russell Mosemann



From Russell.Mosemann at cune.org  Wed Mar 24 04:18:00 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Wed, 24 Mar 2010 06:18:00 -0500
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
Message-ID: <C484C7A362EC491393DDF789D0364BB1@cune.pri>

Joseph L. Casale wrote:

> I have disabled the echo canceller in zt.conf as per the recent thread
> and the results with the mg2 canceller are not very good, is it also
> required to be disabled when using any echo canceller in dahdi?

Yes.

--
Russell Mosemann



From demuel at thephinix.org  Wed Mar 24 04:21:48 2010
From: demuel at thephinix.org (demuel at thephinix.org)
Date: Wed, 24 Mar 2010 11:21:48 -0000
Subject: [Freeswitch-users] Error Compiling FS With Enum Enabled
Message-ID: <3a2b3963e90f85624cdbecc95f854198.squirrel@www.thephinix.org>

Hi,

I got this issue when compiling FS 1.0.5 under FreeBSD 7.0-Stable as show below:

_enum.c  -fPIC -DPIC -o .libs/mod_enum_la-mod_enum.o
mod_enum.c: In function 'enum_lookup':
mod_enum.c:457: error: 'dns_socket' undeclared (first use in this function)
mod_enum.c:457: error: (Each undeclared identifier is reported only once
mod_enum.c:457: error: for each function it appears in.)
mod_enum.c:457: error: expected ';' before 'fd'
cc1: warnings being treated as errors
mod_enum.c:458: warning: ISO C90 forbids mixed declarations and code
mod_enum.c:491: error: 'fd' undeclared (first use in this function)
mod_enum.c: In function 'mod_enum_load':
mod_enum.c:826: error: too few arguments to function 'dns_init'
gmake[4]: *** [mod_enum_la-mod_enum.lo] Error 1
gmake[3]: *** [mod_enum-all] Error 1
gmake[2]: *** [all-recursive] Error 1


Any advice and thanks a lot.

DIB



From velu.technical at gmail.com  Wed Mar 24 04:29:45 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Wed, 24 Mar 2010 16:59:45 +0530
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
Message-ID: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>

Dear all,

    Compiled the wanpipe with freeswitch and configured it.
    When I originate a call from the freeswitch command line it said the
following error and got core dump.

 freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753 &park()
2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
(P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
Version 103  Expecting 100
2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical Error:
PQ Invalid Event lenght from boost rxlen=27 evsz=1027

NOTE:  freeswitch  version is  1.0.trunk ( 17084)
              wanpipe version is 3.5.8.6.
              smg_prid version is 1.63

Please help me.........

Thanks..
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From jcasale at activenetwerx.com  Wed Mar 24 04:36:41 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 11:36:41 +0000
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E6C36@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
	<7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98E6C36@prato.activenetwerx.local>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E6CE4@prato.activenetwerx.local>


>Looking back through my notes I see Anthony Messina suggested I use
>his build script to create the rpms w/ oslec enabled which I just did
>except that the script at http://messinet.com/trac/rpms/browser/tools/svn-build-rpm
>doesn?t build dahdi-tools-libs which is required.

Heh, to early in the morning, forgot about dahdi-tools...

From jcasale at activenetwerx.com  Wed Mar 24 04:37:41 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 11:37:41 +0000
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <7790F777BD4A41B28FCC6DC1A4A7CE2F@cune.pri>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local><7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>
	<7790F777BD4A41B28FCC6DC1A4A7CE2F@cune.pri>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E6CEB@prato.activenetwerx.local>

>We're using OSLEC with DAHDI compiled from source. Follow the directions on the OSLEC web page for DAHDI. We haven't tested it extensively, because our system is just for playing around, but it seems to >be working fine. You will, of course, get the best results from a good card with hardware echo cancellation.

Ok, I was just hoping to use packages. I'll try this if digiums repo doesn't
come back online soon, its down atm:)


From maciej.aniserowicz at gmail.com  Wed Mar 24 06:14:35 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 24 Mar 2010 05:14:35 -0800 (PST)
Subject: [Freeswitch-users] When leg on hold hangs up,
	another leg is hanged up too
Message-ID: <1269436475677-4790809.post@n2.nabble.com>


Hello, it's me again,

I noticed then these steps:

1) dial 2 calls

bgapi originate {ignore_early_media=true}sofia/gateway/agent_gateway/2500
&park()
bgapi originate {ignore_early_media=true}sofia/gateway/g2/2501 &park()

2) bridge them

bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline

3) put the 2nd call on hold:

uuid_hold a85fd47e-051f-42ea-987e-f4e090c7ae78

4) hangup the 2nd call

uuid_kill a85fd47e-051f-42ea-987e-f4e090c7ae78



cause the 1st call to be hanged up as well.

I want it to be parked in this situation.

When I set the [park_after_bridge=true] var in dialstring, it works just
like I want it to.
Is setting this variable the proper way of accomplishing it - keeping
channel alive when the other leg put on hold hangs up?

Thanks,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/When-leg-on-hold-hangs-up-another-leg-is-hanged-up-too-tp4790809p4790809.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rupa at rupa.com  Wed Mar 24 07:57:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 24 Mar 2010 09:57:35 -0500
Subject: [Freeswitch-users] When leg on hold hangs up,
	another leg is 	hanged up too
In-Reply-To: <1269436475677-4790809.post@n2.nabble.com>
References: <1269436475677-4790809.post@n2.nabble.com>
Message-ID: <b39273531003240757l150d495dq17f624705f724037@mail.gmail.com>

since you are controlling the call process, why not just use uuid_park?

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_park

On Wed, Mar 24, 2010 at 8:14 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> Hello, it's me again,
>
> I noticed then these steps:
>
> 1) dial 2 calls
>
> bgapi originate {ignore_early_media=true}sofia/gateway/agent_gateway/2500
> &park()
> bgapi originate {ignore_early_media=true}sofia/gateway/g2/2501 &park()
>
> 2) bridge them
>
> bgapi uuid_transfer f0c0db16-f00f-4b0b-8a00-d3b108d121b2
> intercept:a85fd47e-051f-42ea-987e-f4e090c7ae78,park inline
>
> 3) put the 2nd call on hold:
>
> uuid_hold a85fd47e-051f-42ea-987e-f4e090c7ae78
>
> 4) hangup the 2nd call
>
> uuid_kill a85fd47e-051f-42ea-987e-f4e090c7ae78
>
>
>
> cause the 1st call to be hanged up as well.
>
> I want it to be parked in this situation.
>
> When I set the [park_after_bridge=true] var in dialstring, it works just
> like I want it to.
> Is setting this variable the proper way of accomplishing it - keeping
> channel alive when the other leg put on hold hangs up?
>
> Thanks,
> Maciej Aniserowicz
> --
> View this message in context:
> http://n2.nabble.com/When-leg-on-hold-hangs-up-another-leg-is-hanged-up-too-tp4790809p4790809.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From anthony.minessale at gmail.com  Wed Mar 24 07:58:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Mar 2010 09:58:53 -0500
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
References: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
Message-ID: <191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>

you need the newest beta revision of wanpipe to use with latest trunk

On Wed, Mar 24, 2010 at 6:29 AM, velusamy Krishnan <velu.technical at gmail.com
> wrote:

> Dear all,
>
>     Compiled the wanpipe with freeswitch and configured it.
>     When I originate a call from the freeswitch command line it said the
> following error and got core dump.
>
>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753 &park()
> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
> Version 103  Expecting 100
> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>
> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>               wanpipe version is 3.5.8.6.
>               smg_prid version is 1.63
>
> Please help me.........
>
> Thanks..
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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pstn:+19193869900
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From infos at madovsky.org  Wed Mar 24 08:04:56 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Mar 2010 11:04:56 -0400
Subject: [Freeswitch-users] bypass_media
Message-ID: <D1D7F095DCBE427CAF63FE790AB85B3A@MOBILEE1705>

Hi,

when I use bypass_media on a dialplan extension to transit audio/video
I noticed chunky audio like there is not enough buffer or something like this.
Any idea ?

Thanks
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From brian at freeswitch.org  Wed Mar 24 08:08:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Mar 2010 10:08:13 -0500
Subject: [Freeswitch-users] bypass_media
In-Reply-To: <D1D7F095DCBE427CAF63FE790AB85B3A@MOBILEE1705>
References: <D1D7F095DCBE427CAF63FE790AB85B3A@MOBILEE1705>
Message-ID: <AED6D053-4152-423F-A03C-1068DBD77386@freeswitch.org>

The media is going point to point between the endpoints and FS had nothing to do with it.

/b

On Mar 24, 2010, at 10:04 AM, Madovsky wrote:

> Hi,
>  
> when I use bypass_media on a dialplan extension to transit audio/video
> I noticed chunky audio like there is not enough buffer or something like this.
> Any idea ?
>  
> Thanks

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From anthony.minessale at gmail.com  Wed Mar 24 08:13:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Mar 2010 10:13:17 -0500
Subject: [Freeswitch-users] bypass_media
In-Reply-To: <D1D7F095DCBE427CAF63FE790AB85B3A@MOBILEE1705>
References: <D1D7F095DCBE427CAF63FE790AB85B3A@MOBILEE1705>
Message-ID: <191c3a031003240813v1615e038hf5855045b8632b@mail.gmail.com>

bypass_media takes the media away from FS so any audio problems are between
the 2 devices directly.


On Wed, Mar 24, 2010 at 10:04 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi,
>
> when I use bypass_media on a dialplan extension to transit audio/video
> I noticed chunky audio like there is not enough buffer or something like
> this.
> Any idea ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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pstn:+19193869900
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From jcasale at activenetwerx.com  Wed Mar 24 08:23:42 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 15:23:42 +0000
Subject: [Freeswitch-users] ringback
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E7054@prato.activenetwerx.local>

To answer a call and provide a ringtone during the transfer (while the destination
ua rings and awaits answering), do I simply need to:

<action application="set" data="transfer_ringback="$${us-ring}"/>
<action application="answer"/>
<action application="transfer" data="..."/>

Thanks,
jlc


From maciej.aniserowicz at gmail.com  Wed Mar 24 08:25:25 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Wed, 24 Mar 2010 07:25:25 -0800 (PST)
Subject: [Freeswitch-users] When leg on hold hangs up,
	another leg is 	hanged up too
In-Reply-To: <b39273531003240757l150d495dq17f624705f724037@mail.gmail.com>
References: <1269436475677-4790809.post@n2.nabble.com>
	<b39273531003240757l150d495dq17f624705f724037@mail.gmail.com>
Message-ID: <1269444325694-4791688.post@n2.nabble.com>


The 2nd call can be hanged up manually too, not only from my app by
uuid_kill. I use the uuid_kill scenario just for tests, to see how it
behaves.
-- 
View this message in context: http://n2.nabble.com/When-leg-on-hold-hangs-up-another-leg-is-hanged-up-too-tp4790809p4791688.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Mar 24 08:28:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Mar 2010 10:28:25 -0500
Subject: [Freeswitch-users] Unable to Bridge to FIFO offhook agent
In-Reply-To: <a44688a1003240024i1f5b4dc9jc74f7d3f570b0851@mail.gmail.com>
References: <a44688a1003231726l2eeceab7lcf3b1bc9e12219eb@mail.gmail.com>
	<87f2f3b91003231748n2650ebcdp1253d571c3aee25b@mail.gmail.com>
	<a44688a1003231846w4a63b39eqe3cac4f8b608dcca@mail.gmail.com>
	<87f2f3b91003232216t64253560uf49984a94d485ee@mail.gmail.com>
	<a44688a1003240024i1f5b4dc9jc74f7d3f570b0851@mail.gmail.com>
Message-ID: <191c3a031003240828q5019d01etc432f92bf015ca61@mail.gmail.com>

what is ivr_handler() being executed right before?
it would be relevant to know what that is doing as well.


On Wed, Mar 24, 2010 at 2:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi Michael,
>
> Thanks in advance for all your help.
>
> I pastebin the SIP log as well at http://pastebin.freeswitch.org/
>
> It does not seem to tell much.
>
> Best Regards,
> Woody
>
> On Wed, Mar 24, 2010 at 1:16 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Tue, Mar 23, 2010 at 6:46 PM, Woody Dickson <woodydickson at gmail.com>
> > wrote:
> >>
> >> Hello,
> >>
> >> Here is the log: http://pastebin.freeswitch.org/12533
> >>
> >> I pastebin the relevant debug log.  The highlighted line shows:
> >>
> >> #
> >> 2010-03-24 09:34:00.673468 [WARNING] switch_ivr_bridge.c:1233 Bridge
> >> Failed
> >> sofia/internal/sip:200003 at 192.168.1.13:9000->sofia/internal/
> 200002 at 192.168.1.102
> >>
> >> Please help.
> >>
> >> Best Regards,
> >> Woody
> >>
> > Yeah, the debug log didn't really say why the bridge failed. I'd try a
> few
> > things: pastebin your relevant dialplan bits as well as whatever code you
> > use to put the agents into the queue as members. Also, turn on SIP
> debugging
> > at the console:
> > sofia profile internal siptrace on
> >
> > Look to see if there's anything interesting or unusual in the SIP dialog.
> > You can pastebin that as well.
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From brian at freeswitch.org  Wed Mar 24 08:32:22 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Mar 2010 10:32:22 -0500
Subject: [Freeswitch-users] ringback
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E7054@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E7054@prato.activenetwerx.local>
Message-ID: <68910599-6E3C-4E90-A2F9-5609D098EE87@freeswitch.org>

That all depends it should work but what are you transferring to?

/b

On Mar 24, 2010, at 10:23 AM, Joseph L. Casale wrote:

> <action application="set" data="transfer_ringback="$${us-ring}"/>
> <action application="answer"/>
> <action application="transfer" data="..."/>



From anthony.minessale at gmail.com  Wed Mar 24 08:36:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Mar 2010 10:36:52 -0500
Subject: [Freeswitch-users] When leg on hold hangs up,
	another leg is 	hanged up too
In-Reply-To: <1269444325694-4791688.post@n2.nabble.com>
References: <1269436475677-4790809.post@n2.nabble.com>
	<b39273531003240757l150d495dq17f624705f724037@mail.gmail.com>
	<1269444325694-4791688.post@n2.nabble.com>
Message-ID: <191c3a031003240836r60efa8a4n9cc2fc1ddc17abdb@mail.gmail.com>

either way is fine.

when you hangup up a bridged call the A leg returns to the execute state and
since there are no more apps to execute it hangs up (desired default
behavior) setting park_after_bridge was designed for your desired result of
the call returning to a park state to await further instructions.


On Wed, Mar 24, 2010 at 10:25 AM, Maciej Aniserowicz <
maciej.aniserowicz at gmail.com> wrote:

>
> The 2nd call can be hanged up manually too, not only from my app by
> uuid_kill. I use the uuid_kill scenario just for tests, to see how it
> behaves.
> --
> View this message in context:
> http://n2.nabble.com/When-leg-on-hold-hangs-up-another-leg-is-hanged-up-too-tp4790809p4791688.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

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sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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From thangappan143 at gmail.com  Wed Mar 24 01:33:30 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Wed, 24 Mar 2010 14:03:30 +0530
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
Message-ID: <7aa29e791003240133y4dc95a77r69a26f12b9b2f8f6@mail.gmail.com>

Dear all,

    Compiled the wanpipe with freeswitch and configured it.
    When I originate a call from the freeswitch command line it said the
following error and got core dump.

 freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753 &park()
2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
(P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
Version 103  Expecting 100
2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical Error:
PQ Invalid Event lenght from boost rxlen=27 evsz=1027

NOTE:  freeswitch  version is  1.0.trunk ( 17084)
              wanpipe version is 3.5.8.6.
              smg_prid version is 1.63

Please help me.........

-- 
Regards,
Thangappan.M
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From jcasale at activenetwerx.com  Wed Mar 24 08:42:39 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Wed, 24 Mar 2010 15:42:39 +0000
Subject: [Freeswitch-users] ringback
In-Reply-To: <68910599-6E3C-4E90-A2F9-5609D098EE87@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E7054@prato.activenetwerx.local>
	<68910599-6E3C-4E90-A2F9-5609D098EE87@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98E71D3@prato.activenetwerx.local>

>That all depends it should work but what are you transferring to?

Well, since my tdm card is working now I have an issue whereby our telco plan
prevents me from adjusting the call forward timeout (as suggested earlier) so
I have to answer the zap line before I attempt to transfer it to the ring group.

Thanks!
jlc


From anthony.minessale at gmail.com  Wed Mar 24 08:44:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Mar 2010 10:44:19 -0500
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <7aa29e791003240133y4dc95a77r69a26f12b9b2f8f6@mail.gmail.com>
References: <7aa29e791003240133y4dc95a77r69a26f12b9b2f8f6@mail.gmail.com>
Message-ID: <191c3a031003240844p20d4cfadrd0c907b7a6eb6bf9@mail.gmail.com>

Did you not already send this message with another ID?

You need the latest beta wanpipe driver and you need only 1 email address on
this list.



On Wed, Mar 24, 2010 at 3:33 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>     Compiled the wanpipe with freeswitch and configured it.
>     When I originate a call from the freeswitch command line it said the
> following error and got core dump.
>
>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753 &park()
> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
> Version 103  Expecting 100
> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>
> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>               wanpipe version is 3.5.8.6.
>               smg_prid version is 1.63
>
> Please help me.........
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From christian.loeschenkohl at xpirio.com  Wed Mar 24 10:25:18 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Wed, 24 Mar 2010 18:25:18 +0100
Subject: [Freeswitch-users] donating german prompts
In-Reply-To: <87f2f3b91003231548h7183c96eu4a093f6c441fd042@mail.gmail.com>
References: <4BA934C2.3050209@xpirio.com>
	<87f2f3b91003231548h7183c96eu4a093f6c441fd042@mail.gmail.com>
Message-ID: <4BAA4AFE.3050701@xpirio.com>

hi

thank you
i try to get all the prompts together and then it will go into production (another user - steve hill - did
sent me a full list and i will have to go through this first).
please listen to the voice we got and commtent on this.

mp3 of the voice ist at
http://freeswitch.xpirio.com/

br

Michael Collins wrote:

> Christian,
> 
> This is fantastic! Thank you very much for giving back to the community. 
> We definitely appreciate it.
> -MC
> 
> 2010/3/23 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hello
> 
>     we are going to produce a full set of german audio prompts and
>     donate them to this wonderful project.
>     i have attached a prompt list with the phrases we are going to record.
> 
>     the source of this prompt list is the file
>     docs/phrase/phrase_de.xml. i have also compared it with
>     the english file under docs/phrase/phrase_en.xml an merged this two
>     + translated what was needed/had
>     not been present in the german langunge file from english to german.
> 
>     i would be happy if someone could give me feedback on this and at
>     best review the phrase list in terms
>     of correctness and completeness.
> 
>     br
> 
> 
>     -- 
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From fraserredmond at gmail.com  Wed Mar 24 13:34:03 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 24 Mar 2010 20:34:03 +0000
Subject: [Freeswitch-users] Javascript and Events
Message-ID: <c32ee1cf1003241334i3a03277n47307638d6a42eb7@mail.gmail.com>

I'm trying to get me head round events in Javascript. Looking at this page:
http://wiki.freeswitch.org/wiki/Javascript_Event
...I'd thought I should be able to take the first example and run it in a
call, so that it's listening for all(?) events. Then take the second example
and fire an event.

But nothing is happening. I had a good read through this thread from a year
ago:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/038246.html
...but it seemed to say the resolution was in the event-listener.

In the listener code I changed this:
    if (type == "event") {
            console_log("notice", "Event: " + event_obj.serialize() + "\n");

    }
...to this:
    console_log("notice", "something: " + type + "\n");

And nothing comes up at all (unless I press a key for "dtmf")

Does the javascript event listener only listen for events on it's own
session/channel somehow? I'd thought about adding something like this to the
event-firer:
    e.addHeader("Unique-ID", "8e187c47-da72-471e-82af-adab0bae9347");

Cheers,
Fraser
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From kenfulmer at icstechnologysolutions.com  Wed Mar 24 13:59:36 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Wed, 24 Mar 2010 15:59:36 -0500
Subject: [Freeswitch-users] Cannot lock pid
Message-ID: <00c901cacb94$e9e92710$bdbb7530$@com>

Cannot lock pid file /opt/freeswitch/run/freeswitch.pid.

 

We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
server. Has anyone seen this before?

 

Thanks,

 

Ken Fulmer

 

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From Russell.Mosemann at cune.org  Wed Mar 24 15:00:27 2010
From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org)
Date: Wed, 24 Mar 2010 22:00:27 -0000
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <00c901cacb94$e9e92710$bdbb7530$@com>
Message-ID: <20100324220027.722BF31B5F8@cuneorg-email.cune.pri>

Ken Fulmer <kenfulmer at icstechnologysolutions.com> said:

> We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
> server. Has anyone seen this before?

It looks like a permission problem. Can freeswitch access the path to

/what/ever/freeswitch/log/freeswitch.pid

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!



From msc at freeswitch.org  Wed Mar 24 15:15:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Mar 2010 15:15:41 -0700
Subject: [Freeswitch-users] Javascript and Events
In-Reply-To: <c32ee1cf1003241334i3a03277n47307638d6a42eb7@mail.gmail.com>
References: <c32ee1cf1003241334i3a03277n47307638d6a42eb7@mail.gmail.com>
Message-ID: <87f2f3b91003241515q1cb5349bw57ff6b4f9c2a8c3e@mail.gmail.com>

Is this using event socket? Pastebin your script so we can take a look...
-MC

On Wed, Mar 24, 2010 at 1:34 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I'm trying to get me head round events in Javascript. Looking at this page:
> http://wiki.freeswitch.org/wiki/Javascript_Event
> ...I'd thought I should be able to take the first example and run it in a
> call, so that it's listening for all(?) events. Then take the second example
> and fire an event.
>
> But nothing is happening. I had a good read through this thread from a year
> ago:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/038246.html
> ...but it seemed to say the resolution was in the event-listener.
>
> In the listener code I changed this:
>     if (type == "event") {
>             console_log("notice", "Event: " + event_obj.serialize() +
> "\n");
>     }
> ...to this:
>     console_log("notice", "something: " + type + "\n");
>
> And nothing comes up at all (unless I press a key for "dtmf")
>
> Does the javascript event listener only listen for events on it's own
> session/channel somehow? I'd thought about adding something like this to the
> event-firer:
>     e.addHeader("Unique-ID", "8e187c47-da72-471e-82af-adab0bae9347");
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Wed Mar 24 15:44:45 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 25 Mar 2010 06:44:45 +0800
Subject: [Freeswitch-users] Success with a TDM410P finally
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98E66FF@prato.activenetwerx.local>
	<7d0bfd8c1003232203h2af9fe64reeb5e4e5ae8810f6@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98E6BD7@prato.activenetwerx.local>
Message-ID: <7d0bfd8c1003241544t4e59054an2d07fa534d054dcc@mail.gmail.com>

hi joseph,

the TDM400P is in the back burner for the meantime. i shifted to gateways. i
was using digium zaptel source. it was working w/ oslec. i just had problems
in the call progress tone detection for Philippine settings. i think i
should try again using the dahdi package.

-nandy
On Wed, Mar 24, 2010 at 6:31 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >hi joseph,
> >
> >have u tested the Oslec echo-canceller?
>
> Nandy,
> I'd love to, but I have never had any luck getting it to work.
> Currently, I am using dahdi packages from digiums yum repo. Are
> you using rpm's or compiling from source?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wchao at yahoo.com  Wed Mar 24 18:35:06 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Wed, 24 Mar 2010 21:35:06 -0400 (EDT)
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys phones
Message-ID: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>

I am using Freeswitch and I have some phones that spontaneously reboot 
(often in the middle of a call).

I did some searching around on the web, but didn't find any information 
that really applied to the problem I'm experiencing. I have a few dozen 
Polycom phones and a handful of Linksys phones of various models: Polycom 
IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921, 
SPA941.

I also have snom phones, but they don't reboot spontaneously. It's only 
the Polycom and Linksys phones that reboot spontaneously, and it seems to 
be correlated to call volume. Heavy usage of the Polycom and Linksys 
phones causes them to reboot more frequently (i.e. shorter duration 
between the reboots). For people making 50 calls a day, the Polycom phones 
spontaneously reboot every three days or so (and always on a call, it 
seems). For people making one or two calls a day, the Polycom will go 
months without a spontaneous reboot. This correlation between call volume 
and reboot frequency leads me to believe that there is some bug in the 
Polycom firmware, perhaps some of overflow condition that accumulates over 
time and eventually causes a crash (and subsequent reboot).

I was wondering if anybody else experienced something similar, and whether 
anyone had developed a solution. One thing I am considering is to set up a 
script to reboot the Polycom phones in the middle of the night every day, 
which would hopefully prevent them from spontaneously rebooting during the 
day by preventing the phones from ever reaching the overflow condition (if 
that is indeed what causes the reboots). A nightly reboot would be a 
hackish way of fixing the problem, so I'd also love to hear if anybody has 
any other speculations on what might be causing the spontaneous reboots. 
We have phones in multiple buildings, and we've discovered that the 
spontaneous reboots happen on phones on different networks, so it's not 
specific to a particular LAN segment. For the most part, we are using the 
standard config files for the Polycom phones and for the Linksys phones.


From brian at freeswitch.org  Wed Mar 24 18:44:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Mar 2010 20:44:55 -0500
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
	phones
In-Reply-To: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
Message-ID: <30663396-E364-47C9-95E8-F14BA773497E@freeswitch.org>

I haven't... and I use the shit out of my polycom's, I use them for 10 hours or more solid a day on a single call.. while the 670 on my right makes 30-40 calls or more a day depending on what I'm doing.  Now you failed to tell us what Firmware Rev you're running and what options you have turned on?

/b

On Mar 24, 2010, at 8:35 PM, Wellie Chao wrote:

> I am using Freeswitch and I have some phones that spontaneously reboot 
> (often in the middle of a call).
> 
> I did some searching around on the web, but didn't find any information 
> that really applied to the problem I'm experiencing. I have a few dozen 
> Polycom phones and a handful of Linksys phones of various models: Polycom 
> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921, 
> SPA941.
> 
> I also have snom phones, but they don't reboot spontaneously. It's only 
> the Polycom and Linksys phones that reboot spontaneously, and it seems to 
> be correlated to call volume. Heavy usage of the Polycom and Linksys 
> phones causes them to reboot more frequently (i.e. shorter duration 
> between the reboots). For people making 50 calls a day, the Polycom phones 
> spontaneously reboot every three days or so (and always on a call, it 
> seems). For people making one or two calls a day, the Polycom will go 
> months without a spontaneous reboot. This correlation between call volume 
> and reboot frequency leads me to believe that there is some bug in the 
> Polycom firmware, perhaps some of overflow condition that accumulates over 
> time and eventually causes a crash (and subsequent reboot).
> 
> I was wondering if anybody else experienced something similar, and whether 
> anyone had developed a solution. One thing I am considering is to set up a 
> script to reboot the Polycom phones in the middle of the night every day, 
> which would hopefully prevent them from spontaneously rebooting during the 
> day by preventing the phones from ever reaching the overflow condition (if 
> that is indeed what causes the reboots). A nightly reboot would be a 
> hackish way of fixing the problem, so I'd also love to hear if anybody has 
> any other speculations on what might be causing the spontaneous reboots. 
> We have phones in multiple buildings, and we've discovered that the 
> spontaneous reboots happen on phones on different networks, so it's not 
> specific to a particular LAN segment. For the most part, we are using the 
> standard config files for the Polycom phones and for the Linksys phones.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chris at fowler.cc  Wed Mar 24 19:59:11 2010
From: chris at fowler.cc (Chris Fowler)
Date: Wed, 24 Mar 2010 22:59:11 -0400
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and
	Linksys	phones
In-Reply-To: <30663396-E364-47C9-95E8-F14BA773497E@freeswitch.org>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<30663396-E364-47C9-95E8-F14BA773497E@freeswitch.org>
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C7225BCC0@VMBX113.ihostexchange.net>

I have ~100 Polycom's installed and never see unexplained or spontaneous reboots.  

I do see them reboot if the switch they are connected to momentarily drops the link (e.g. if the switch reboots) (to be clear this is with non-PoE powered phones - as PoE powered phones would of course reboot if the switch reboots...).

I'd suggest upgrading to the latest bootrom and firmware for the Polycoms.  They also upload their logs when they reboot (to the provisioning server) - so take a peek in there for clues.

Chris.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, March 24, 2010 6:45 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys phones

I haven't... and I use the shit out of my polycom's, I use them for 10 hours or more solid a day on a single call.. while the 670 on my right makes 30-40 calls or more a day depending on what I'm doing.  Now you failed to tell us what Firmware Rev you're running and what options you have turned on?

/b

On Mar 24, 2010, at 8:35 PM, Wellie Chao wrote:

> I am using Freeswitch and I have some phones that spontaneously reboot
> (often in the middle of a call).
>
> I did some searching around on the web, but didn't find any information
> that really applied to the problem I'm experiencing. I have a few dozen
> Polycom phones and a handful of Linksys phones of various models: Polycom
> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921,
> SPA941.
>
> I also have snom phones, but they don't reboot spontaneously. It's only
> the Polycom and Linksys phones that reboot spontaneously, and it seems to
> be correlated to call volume. Heavy usage of the Polycom and Linksys
> phones causes them to reboot more frequently (i.e. shorter duration
> between the reboots). For people making 50 calls a day, the Polycom phones
> spontaneously reboot every three days or so (and always on a call, it
> seems). For people making one or two calls a day, the Polycom will go
> months without a spontaneous reboot. This correlation between call volume
> and reboot frequency leads me to believe that there is some bug in the
> Polycom firmware, perhaps some of overflow condition that accumulates over
> time and eventually causes a crash (and subsequent reboot).
>
> I was wondering if anybody else experienced something similar, and whether
> anyone had developed a solution. One thing I am considering is to set up a
> script to reboot the Polycom phones in the middle of the night every day,
> which would hopefully prevent them from spontaneously rebooting during the
> day by preventing the phones from ever reaching the overflow condition (if
> that is indeed what causes the reboots). A nightly reboot would be a
> hackish way of fixing the problem, so I'd also love to hear if anybody has
> any other speculations on what might be causing the spontaneous reboots.
> We have phones in multiple buildings, and we've discovered that the
> spontaneous reboots happen on phones on different networks, so it's not
> specific to a particular LAN segment. For the most part, we are using the
> standard config files for the Polycom phones and for the Linksys phones.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From infos at madovsky.org  Wed Mar 24 20:28:48 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 24 Mar 2010 23:28:48 -0400
Subject: [Freeswitch-users] hangup leg B from registered on another FS
Message-ID: <FAE674A59EDD49E2B2EA7F6205074A73@MOBILEE1705>

Hi,

maybe it's my mistake:
2 FS running with a shared DB, same sip profile.
if leg A registered on FS 1 call leg B registered on FS 2
and hangs up, leg B stays onlline even after rtp time out.
also every registration take a row for every node in sip_registration table, is it ok ?

Thanks
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From gkuri at ieee.org  Wed Mar 24 20:42:38 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 24 Mar 2010 20:42:38 -0700
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
	phones
In-Reply-To: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
Message-ID: <8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>

If you're using central provisioning with the Linksys/Cisco Small
Business series phones and you leave the Message_Waiting option in the
config (which is the default, very stupid), the phones will reboot
every time they sync the config, if they have a voicemail waiting.

Check to see if the Linksys spontaneous reboots correlate with pulling
down the config (assuming you're using central provisioning) and make
sure you either comment or delete Message_Waiting from the config.

You may also want to see if there's any correlation with DHCP and the
phones renewing their IP. We put all of our phones on DHCP
reservations, so their IP doesn't/shouldn't change. I'm not sure what
the phone will do if it goes to renew its address and the DHCP server
hands it a different IP, it may cause the phone to reboot?

Cheers,
Gabe


On Wed, Mar 24, 2010 at 6:35 PM, Wellie Chao <wchao at yahoo.com> wrote:
> I am using Freeswitch and I have some phones that spontaneously reboot
> (often in the middle of a call).
>
> I did some searching around on the web, but didn't find any information
> that really applied to the problem I'm experiencing. I have a few dozen
> Polycom phones and a handful of Linksys phones of various models: Polycom
> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921,
> SPA941.
>
> I also have snom phones, but they don't reboot spontaneously. It's only
> the Polycom and Linksys phones that reboot spontaneously, and it seems to
> be correlated to call volume. Heavy usage of the Polycom and Linksys
> phones causes them to reboot more frequently (i.e. shorter duration
> between the reboots). For people making 50 calls a day, the Polycom phones
> spontaneously reboot every three days or so (and always on a call, it
> seems). For people making one or two calls a day, the Polycom will go
> months without a spontaneous reboot. This correlation between call volume
> and reboot frequency leads me to believe that there is some bug in the
> Polycom firmware, perhaps some of overflow condition that accumulates over
> time and eventually causes a crash (and subsequent reboot).
>
> I was wondering if anybody else experienced something similar, and whether
> anyone had developed a solution. One thing I am considering is to set up a
> script to reboot the Polycom phones in the middle of the night every day,
> which would hopefully prevent them from spontaneously rebooting during the
> day by preventing the phones from ever reaching the overflow condition (if
> that is indeed what causes the reboots). A nightly reboot would be a
> hackish way of fixing the problem, so I'd also love to hear if anybody has
> any other speculations on what might be causing the spontaneous reboots.
> We have phones in multiple buildings, and we've discovered that the
> spontaneous reboots happen on phones on different networks, so it's not
> specific to a particular LAN segment. For the most part, we are using the
> standard config files for the Polycom phones and for the Linksys phones.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From spencer at 5ninesolutions.com  Wed Mar 24 21:28:48 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 21:28:48 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
Message-ID: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>

Hello all,
I use Bandtel as one of my trunking providers and Freeswitch will work  
flawlessly for 24-36 hours and then stop responding to SIP requests.   
I have monit setup to send an options ping once a minute and restart  
the process if it doesn't get a reply.  This has been working ok but  
the Freeswitch process has to get restarted about once a day on  
average.  Has anybody else had any issues like this?  They are using  
fairly standard issue stuff, Freeswitch registers to SER 0.9.6 and  
they are using Sippy as a B2BUA on their end.  The log shows a call  
ending, a period of no activity, and Freeswitch attempting to register  
repeatedly, the process then stops responding to requests and monit  
restarts Freeswitch.

2010-03-24 17:31:03.897756 [DEBUG] switch_core_state_machine.c:439 (sofia/default/13105106983 at 208.70.78.37 
) State DESTROY going to sleep
2010-03-24 17:33:16.353544 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-24 17:37:58.452249 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-24 17:42:40.588167 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-24 17:47:22.529220 [NOTICE] sofia_reg.c:336 Registering bandtel
2010-03-24 17:53:15.899570 [INFO] switch_core.c:1734 Shutting down

Here is a typical register request.
   
------------------------------------------------------------------------
2010-03-24 21:12:59.955693 [NOTICE] sofia_reg.c:336 Registering bandtel
send 772 bytes to udp/[66.237.65.67]:5060 at 04:12:59.956245:
     
------------------------------------------------------------------------
    REGISTER sip:66.237.65.67 SIP/2.0
    Via: SIP/2.0/UDP 208.70.78.36;rport;branch=z9hG4bKZcD7H9aN1D50j
    Max-Forwards: 70
    From: <sip: 
201009000001 at 66.237.65.67;transport=udp>;tag=3yZ6ZSZUFj3gQ
    To: <sip:201009000001 at 66.237.65.67;transport=udp>
    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
    CSeq: 128622152 REGISTER
    Contact: <sip:gw+bandtel at 208.70.78.36:5060;transport=udp;gw=bandtel>
    Expires: 390
    User-Agent: FreeSWITCH
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
REGISTER, NOTIFY
    Supported: timer, precondition, path, replaces
    Authorization: Digest username="201009000001", realm="10.1.2.1",  
nonce="4baabf0e1e68a1fdfbde46913ca850da0cbecc8a", algorithm=MD5, uri="sip:66.237.65.67 
", response="3974689c5cb3eeb4f09e634716d2239e"
    Content-Length: 0

     
------------------------------------------------------------------------
recv 505 bytes from udp/[66.237.65.67]:5060 at 04:12:59.980247:
     
------------------------------------------------------------------------
    SIP/2.0 200 OK
    From: <sip: 
201009000001 at 66.237.65.67;transport=UDP>;tag=3yZ6ZSZUFj3gQ
    To: <sip: 
201009000001 
@66.237.65.67;transport=UDP>;tag=2e41cc73d384c8f458f34dd410d7744b-b5e6
    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
    CSeq: 128622152 REGISTER
    Via: SIP/2.0/UDP 208.70.78.36;rport=5060;branch=z9hG4bKZcD7H9aN1D50j
    PortaBilling: currency:USD
    Server: Sip EXpress router (0.9.6 (i386/freebsd))
    Contact: <sip:gw+bandtel at 66.237.65.67:5060;gw=bandtel>;expires=385
    Expires: 385
    Content-Length: 0

     
------------------------------------------------------------------------
2010-03-24 21:12:59.979821 [DEBUG] sofia_reg.c:1478 Changing expire  
time to 385 by request of proxy sip:66.237.65.67

What is the best way to debug this?  It seems strange that everything  
works for a while and then stops.

Thanks,
Spencer

--
Spencer Thomason
5Nine Solutions LLC
e. sales at 5ninesolutions.com
p. +1.888.271.7959
f. +1.310.510.6980



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From brian at freeswitch.org  Wed Mar 24 21:34:06 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Mar 2010 23:34:06 -0500
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
Message-ID: <5934EA30-6690-4E39-9C16-BF5E1AAF00E1@freeswitch.org>

are you on SVN trunk?

/b

On Mar 24, 2010, at 11:28 PM, Spencer Thomason wrote:

> Hello all,
> I use Bandtel as one of my trunking providers and Freeswitch will work flawlessly for 24-36 hours and then stop responding to SIP requests.  I have monit setup to send an options ping once a minute and restart the process if it doesn't get a reply.  This has been working ok but the Freeswitch process has to get restarted about once a day on average.  Has anybody else had any issues like this?  They are using fairly standard issue stuff, Freeswitch registers to SER 0.9.6 and they are using Sippy as a B2BUA on their end.  The log shows a call ending, a period of no activity, and Freeswitch attempting to register repeatedly, the process then stops responding to requests and monit restarts Freeswitch.
> 
> 2010-03-24 17:31:03.897756 [DEBUG] switch_core_state_machine.c:439 (sofia/default/13105106983 at 208.70.78.37) State DESTROY going to sleep
> 2010-03-24 17:33:16.353544 [NOTICE] sofia_reg.c:336 Registering bandtel
> 2010-03-24 17:37:58.452249 [NOTICE] sofia_reg.c:336 Registering bandtel
> 2010-03-24 17:42:40.588167 [NOTICE] sofia_reg.c:336 Registering bandtel
> 2010-03-24 17:47:22.529220 [NOTICE] sofia_reg.c:336 Registering bandtel
> 2010-03-24 17:53:15.899570 [INFO] switch_core.c:1734 Shutting down
> 
> Here is a typical register request.
>  ------------------------------------------------------------------------
> 2010-03-24 21:12:59.955693 [NOTICE] sofia_reg.c:336 Registering bandtel
> send 772 bytes to udp/[66.237.65.67]:5060 at 04:12:59.956245:
>    ------------------------------------------------------------------------
>    REGISTER sip:66.237.65.67 SIP/2.0
>    Via: SIP/2.0/UDP 208.70.78.36;rport;branch=z9hG4bKZcD7H9aN1D50j
>    Max-Forwards: 70
>    From: <sip:201009000001 at 66.237.65.67;transport=udp>;tag=3yZ6ZSZUFj3gQ
>    To: <sip:201009000001 at 66.237.65.67;transport=udp>
>    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
>    CSeq: 128622152 REGISTER
>    Contact: <sip:gw+bandtel at 208.70.78.36:5060;transport=udp;gw=bandtel>
>    Expires: 390
>    User-Agent: FreeSWITCH
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Authorization: Digest username="201009000001", realm="10.1.2.1", nonce="4baabf0e1e68a1fdfbde46913ca850da0cbecc8a", algorithm=MD5, uri="sip:66.237.65.67", response="3974689c5cb3eeb4f09e634716d2239e"
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 505 bytes from udp/[66.237.65.67]:5060 at 04:12:59.980247:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    From: <sip:201009000001 at 66.237.65.67;transport=UDP>;tag=3yZ6ZSZUFj3gQ
>    To: <sip:201009000001 at 66.237.65.67;transport=UDP>;tag=2e41cc73d384c8f458f34dd410d7744b-b5e6
>    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
>    CSeq: 128622152 REGISTER
>    Via: SIP/2.0/UDP 208.70.78.36;rport=5060;branch=z9hG4bKZcD7H9aN1D50j
>    PortaBilling: currency:USD
>    Server: Sip EXpress router (0.9.6 (i386/freebsd))
>    Contact: <sip:gw+bandtel at 66.237.65.67:5060;gw=bandtel>;expires=385
>    Expires: 385
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> 2010-03-24 21:12:59.979821 [DEBUG] sofia_reg.c:1478 Changing expire time to 385 by request of proxy sip:66.237.65.67
> 
> What is the best way to debug this?  It seems strange that everything works for a while and then stops.
> 
> Thanks,
> Spencer
> 
> --
> Spencer Thomason
> 5Nine Solutions LLC
> e. sales at 5ninesolutions.com
> p. +1.888.271.7959
> f. +1.310.510.6980
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jasonjgw.net  Wed Mar 24 21:42:00 2010
From: jason at jasonjgw.net (Jason White)
Date: Thu, 25 Mar 2010 15:42:00 +1100
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
Message-ID: <20100325044200.GA7885@jdc.jasonjgw.net>

Spencer Thomason <spencer at 5ninesolutions.com> wrote:
> What is the best way to debug this?  

1. Upgrade to the latest version from the svn repository if you haven't done
so already. You didn't specify which version of FreeSWITCH you were using, but
if it's old, upgrade it and try to reproduce the problem with the latest
version.

2. Assuming you can reproduce the problem with the latest code from svn, I
suppose it's time for gdb - instructions are on the wiki.

Can you run fs_cli and connect to it in this state or does it hang completely?



From mustafa.pk at gmail.com  Wed Mar 24 21:46:13 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Thu, 25 Mar 2010 09:46:13 +0500
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
	phones
In-Reply-To: <8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
Message-ID: <8213d6071003242146vf90a66q9afb4cd04949e222@mail.gmail.com>

Hi,
i had experienced a similar problem with polycom ip500, and this was a power
adapter problem causing phone to reboot when speaker phone is on and sound
volume is high :), replacing 500mp adapter with 1000ma fixed the problem. if
you are using PoE then it's something related to phone firmware definitely.

-m

On Thu, Mar 25, 2010 at 8:42 AM, Gabriel Kuri <gkuri at ieee.org> wrote:

> If you're using central provisioning with the Linksys/Cisco Small
> Business series phones and you leave the Message_Waiting option in the
> config (which is the default, very stupid), the phones will reboot
> every time they sync the config, if they have a voicemail waiting.
>
> Check to see if the Linksys spontaneous reboots correlate with pulling
> down the config (assuming you're using central provisioning) and make
> sure you either comment or delete Message_Waiting from the config.
>
> You may also want to see if there's any correlation with DHCP and the
> phones renewing their IP. We put all of our phones on DHCP
> reservations, so their IP doesn't/shouldn't change. I'm not sure what
> the phone will do if it goes to renew its address and the DHCP server
> hands it a different IP, it may cause the phone to reboot?
>
> Cheers,
> Gabe
>
>
> On Wed, Mar 24, 2010 at 6:35 PM, Wellie Chao <wchao at yahoo.com> wrote:
> > I am using Freeswitch and I have some phones that spontaneously reboot
> > (often in the middle of a call).
> >
> > I did some searching around on the web, but didn't find any information
> > that really applied to the problem I'm experiencing. I have a few dozen
> > Polycom phones and a handful of Linksys phones of various models: Polycom
> > IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921,
> > SPA941.
> >
> > I also have snom phones, but they don't reboot spontaneously. It's only
> > the Polycom and Linksys phones that reboot spontaneously, and it seems to
> > be correlated to call volume. Heavy usage of the Polycom and Linksys
> > phones causes them to reboot more frequently (i.e. shorter duration
> > between the reboots). For people making 50 calls a day, the Polycom
> phones
> > spontaneously reboot every three days or so (and always on a call, it
> > seems). For people making one or two calls a day, the Polycom will go
> > months without a spontaneous reboot. This correlation between call volume
> > and reboot frequency leads me to believe that there is some bug in the
> > Polycom firmware, perhaps some of overflow condition that accumulates
> over
> > time and eventually causes a crash (and subsequent reboot).
> >
> > I was wondering if anybody else experienced something similar, and
> whether
> > anyone had developed a solution. One thing I am considering is to set up
> a
> > script to reboot the Polycom phones in the middle of the night every day,
> > which would hopefully prevent them from spontaneously rebooting during
> the
> > day by preventing the phones from ever reaching the overflow condition
> (if
> > that is indeed what causes the reboots). A nightly reboot would be a
> > hackish way of fixing the problem, so I'd also love to hear if anybody
> has
> > any other speculations on what might be causing the spontaneous reboots.
> > We have phones in multiple buildings, and we've discovered that the
> > spontaneous reboots happen on phones on different networks, so it's not
> > specific to a particular LAN segment. For the most part, we are using the
> > standard config files for the Polycom phones and for the Linksys phones.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From spencer at 5ninesolutions.com  Wed Mar 24 21:46:51 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 21:46:51 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <5934EA30-6690-4E39-9C16-BF5E1AAF00E1@freeswitch.org>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<5934EA30-6690-4E39-9C16-BF5E1AAF00E1@freeswitch.org>
Message-ID: <3C21AD7D-3464-48F6-A205-9ABC46759641@5ninesolutions.com>

Yes.

freeswitch at internal> version
FreeSWITCH Version 1.0.trunk (17097M)

Spencer

On Mar 24, 2010, at 9:34 PM, Brian West wrote:

> are you on SVN trunk?
>
> /b
>
> On Mar 24, 2010, at 11:28 PM, Spencer Thomason wrote:
>
>> Hello all,
>> I use Bandtel as one of my trunking providers and Freeswitch will  
>> work flawlessly for 24-36 hours and then stop responding to SIP  
>> requests.  I have monit setup to send an options ping once a minute  
>> and restart the process if it doesn't get a reply.  This has been  
>> working ok but the Freeswitch process has to get restarted about  
>> once a day on average.  Has anybody else had any issues like this?   
>> They are using fairly standard issue stuff, Freeswitch registers to  
>> SER 0.9.6 and they are using Sippy as a B2BUA on their end.  The  
>> log shows a call ending, a period of no activity, and Freeswitch  
>> attempting to register repeatedly, the process then stops  
>> responding to requests and monit restarts Freeswitch.
>>
>> 2010-03-24 17:31:03.897756 [DEBUG] switch_core_state_machine.c:439 (sofia/default/13105106983 at 208.70.78.37 
>> ) State DESTROY going to sleep
>> 2010-03-24 17:33:16.353544 [NOTICE] sofia_reg.c:336 Registering  
>> bandtel
>> 2010-03-24 17:37:58.452249 [NOTICE] sofia_reg.c:336 Registering  
>> bandtel
>> 2010-03-24 17:42:40.588167 [NOTICE] sofia_reg.c:336 Registering  
>> bandtel
>> 2010-03-24 17:47:22.529220 [NOTICE] sofia_reg.c:336 Registering  
>> bandtel
>> 2010-03-24 17:53:15.899570 [INFO] switch_core.c:1734 Shutting down
>>
>> Here is a typical register request.
>>   
>> ------------------------------------------------------------------------
>> 2010-03-24 21:12:59.955693 [NOTICE] sofia_reg.c:336 Registering  
>> bandtel
>> send 772 bytes to udp/[66.237.65.67]:5060 at 04:12:59.956245:
>>     
>> ------------------------------------------------------------------------
>>    REGISTER sip:66.237.65.67 SIP/2.0
>>    Via: SIP/2.0/UDP 208.70.78.36;rport;branch=z9hG4bKZcD7H9aN1D50j
>>    Max-Forwards: 70
>>    From: <sip: 
>> 201009000001 at 66.237.65.67;transport=udp>;tag=3yZ6ZSZUFj3gQ
>>    To: <sip:201009000001 at 66.237.65.67;transport=udp>
>>    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
>>    CSeq: 128622152 REGISTER
>>    Contact: <sip:gw+bandtel at 208.70.78.36:5060;transport=udp;gw=bandtel 
>> >
>>    Expires: 390
>>    User-Agent: FreeSWITCH
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
>> REGISTER, NOTIFY
>>    Supported: timer, precondition, path, replaces
>>    Authorization: Digest username="201009000001", realm="10.1.2.1",  
>> nonce="4baabf0e1e68a1fdfbde46913ca850da0cbecc8a", algorithm=MD5,  
>> uri="sip:66.237.65.67", response="3974689c5cb3eeb4f09e634716d2239e"
>>    Content-Length: 0
>>
>>     
>> ------------------------------------------------------------------------
>> recv 505 bytes from udp/[66.237.65.67]:5060 at 04:12:59.980247:
>>     
>> ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    From: <sip: 
>> 201009000001 at 66.237.65.67;transport=UDP>;tag=3yZ6ZSZUFj3gQ
>>    To: <sip: 
>> 201009000001 
>> @66.237.65.67;transport=UDP>;tag=2e41cc73d384c8f458f34dd410d7744b- 
>> b5e6
>>    Call-ID: b1eae39a-37ae-11df-9bcf-f1944b4d3175
>>    CSeq: 128622152 REGISTER
>>    Via: SIP/2.0/UDP  
>> 208.70.78.36;rport=5060;branch=z9hG4bKZcD7H9aN1D50j
>>    PortaBilling: currency:USD
>>    Server: Sip EXpress router (0.9.6 (i386/freebsd))
>>    Contact: <sip:gw+bandtel at 66.237.65.67:5060;gw=bandtel>;expires=385
>>    Expires: 385
>>    Content-Length: 0
>>
>>     
>> ------------------------------------------------------------------------
>> 2010-03-24 21:12:59.979821 [DEBUG] sofia_reg.c:1478 Changing expire  
>> time to 385 by request of proxy sip:66.237.65.67
>>
>> What is the best way to debug this?  It seems strange that  
>> everything works for a while and then stops.
>>
>> Thanks,
>> Spencer
>>
>> --
>> Spencer Thomason
>> 5Nine Solutions LLC
>> e. sales at 5ninesolutions.com
>> p. +1.888.271.7959
>> f. +1.310.510.6980
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From velu.technical at gmail.com  Wed Mar 24 21:52:01 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Thu, 25 Mar 2010 10:22:01 +0530
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>
References: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
	<191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>
Message-ID: <1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>

Installed the wanpipe version(wanpipe-3.5.10.10b.tgz) which is suggested in
the http://wiki.sangoma.com/wanpipe-freeswitch-install.

But, the same is happening.
>From sangoma_mgd.log file found the following,

sangoma_prid: = Sangoma PRI Protocol Stack Daemon
sangoma_prid: = Version: 1.75
sangoma_prid: = Date: Mar 18 2010
sangoma_prid: = Wanpipe Release: wanpipe-3.5.10.10
sangoma_prid: = Revision:Revision: 15921

Do you mean some other wanpipe release?

On Wed, Mar 24, 2010 at 8:28 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you need the newest beta revision of wanpipe to use with latest trunk
>
> On Wed, Mar 24, 2010 at 6:29 AM, velusamy Krishnan <
> velu.technical at gmail.com> wrote:
>
>> Dear all,
>>
>>     Compiled the wanpipe with freeswitch and configured it.
>>     When I originate a call from the freeswitch command line it said the
>> following error and got core dump.
>>
>>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753
>> &park()
>> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
>> Version 103  Expecting 100
>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
>> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>>
>> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>>               wanpipe version is 3.5.8.6.
>>               smg_prid version is 1.63
>>
>> Please help me.........
>>
>> Thanks..
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From spencer at 5ninesolutions.com  Wed Mar 24 21:53:56 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 21:53:56 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <20100325044200.GA7885@jdc.jasonjgw.net>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
Message-ID: <8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>

fs_cli will still connect but you cannot issue any commands to sofia,  
i.e. restart, etc

I am on svn trunk.  The problem is I haven't been able to reproduce  
this in the lab, it only happens when it goes into production which  
makes debugging a challenge at best, the process has to get restarted  
as fast as possible.

Spencer

On Mar 24, 2010, at 9:42 PM, Jason White wrote:

> Spencer Thomason <spencer at 5ninesolutions.com> wrote:
>> What is the best way to debug this?
>
> 1. Upgrade to the latest version from the svn repository if you  
> haven't done
> so already. You didn't specify which version of FreeSWITCH you were  
> using, but
> if it's old, upgrade it and try to reproduce the problem with the  
> latest
> version.
>
> 2. Assuming you can reproduce the problem with the latest code from  
> svn, I
> suppose it's time for gdb - instructions are on the wiki.
>
> Can you run fs_cli and connect to it in this state or does it hang  
> completely?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Wed Mar 24 21:58:51 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Mar 2010 23:58:51 -0500
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
Message-ID: <B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>

Can you elaborate a bit more on what is going on?  Maybe you just have build skew .

/b

On Mar 24, 2010, at 11:53 PM, Spencer Thomason wrote:

> fs_cli will still connect but you cannot issue any commands to sofia,  
> i.e. restart, etc
> 
> I am on svn trunk.  The problem is I haven't been able to reproduce  
> this in the lab, it only happens when it goes into production which  
> makes debugging a challenge at best, the process has to get restarted  
> as fast as possible.
> 
> Spencer



From spencer at 5ninesolutions.com  Wed Mar 24 22:18:17 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 22:18:17 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
Message-ID: <5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>

I have rebuilt and rebuilt, once every couple of days.  This has been  
going on for a while.  I was using CentOS but I switched to Ubuntu to  
try to build against some newer libraries.  I thought the problem was  
resolved but it is very intermittent so its hard to tell.  The gdb  
output is at http://pastebin.freeswitch.org/12451  The Freeswitch  
process just seems to randomly start using 100% of the CPU and stop  
responding about once every 12-36 hours.  Today it had to be restarted  
twice, somedays not at all, very light load, about 100 calls per day.

root at sip:/usr/local/freeswitch/bin# cat /proc/cpuinfo
processor	: 0
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2205.000
cache size	: 1024 KB
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 cmov pat clflush mmx fxsr sse sse2 ht  
syscall nx mmxext fxsr_opt lm 3dnowext 3dnow rep_good pni hypervisor  
cmp_legacy
bogomips	: 4410.00
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 1
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2205.000
cache size	: 1024 KB
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 cmov pat clflush mmx fxsr sse sse2 ht  
syscall nx mmxext fxsr_opt lm 3dnowext 3dnow rep_good pni hypervisor  
cmp_legacy
bogomips	: 4410.00
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 2
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2205.000
cache size	: 1024 KB
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 cmov pat clflush mmx fxsr sse sse2 ht  
syscall nx mmxext fxsr_opt lm 3dnowext 3dnow rep_good pni hypervisor  
cmp_legacy
bogomips	: 4410.00
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp

processor	: 3
vendor_id	: AuthenticAMD
cpu family	: 15
model		: 33
model name	: AMD Opteron(tm) Processor 275
stepping	: 2
cpu MHz		: 2205.000
cache size	: 1024 KB
fpu		: yes
fpu_exception	: yes
cpuid level	: 1
wp		: yes
flags		: fpu de tsc msr pae cx8 cmov pat clflush mmx fxsr sse sse2 ht  
syscall nx mmxext fxsr_opt lm 3dnowext 3dnow rep_good pni hypervisor  
cmp_legacy
bogomips	: 4410.00
TLB size	: 1024 4K pages
clflush size	: 64
cache_alignment	: 64
address sizes	: 40 bits physical, 48 bits virtual
power management: ts fid vid ttp



On Mar 24, 2010, at 9:58 PM, Brian West wrote:

> Can you elaborate a bit more on what is going on?  Maybe you just  
> have build skew .
>
> /b
>
> On Mar 24, 2010, at 11:53 PM, Spencer Thomason wrote:
>
>> fs_cli will still connect but you cannot issue any commands to sofia,
>> i.e. restart, etc
>>
>> I am on svn trunk.  The problem is I haven't been able to reproduce
>> this in the lab, it only happens when it goes into production which
>> makes debugging a challenge at best, the process has to get restarted
>> as fast as possible.
>>
>> Spencer
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Wed Mar 24 22:23:19 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 00:23:19 -0500
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
Message-ID: <8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>

Well that was your first mistake.. why did you feel the need to use Ubuntu?  Also what does uname -a say?

/b

On Mar 25, 2010, at 12:18 AM, Spencer Thomason wrote:

> I have rebuilt and rebuilt, once every couple of days.  This has been  
> going on for a while.  I was using CentOS but I switched to Ubuntu to  
> try to build against some newer libraries.  I thought the problem was  
> resolved but it is very intermittent so its hard to tell.  The gdb  
> output is at http://pastebin.freeswitch.org/12451  The Freeswitch  
> process just seems to randomly start using 100% of the CPU and stop  
> responding about once every 12-36 hours.  Today it had to be restarted  
> twice, somedays not at all, very light load, about 100 calls per day.

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From spencer at 5ninesolutions.com  Wed Mar 24 22:37:51 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 22:37:51 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
Message-ID: <D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>

I dunno, just thought I'd try something different.. :-)

root at sip:/usr/local/freeswitch/bin# uname -r
2.6.31-20-generic

I'd much rather use CentOS.  The Ubuntu machine is running under Xen  
3.4.1 if that matters.



On Mar 24, 2010, at 10:23 PM, Brian West wrote:

> Well that was your first mistake.. why did you feel the need to use  
> Ubuntu?  Also what does uname -a say?
>
> /b
>
> On Mar 25, 2010, at 12:18 AM, Spencer Thomason wrote:
>
>> I have rebuilt and rebuilt, once every couple of days.  This has been
>> going on for a while.  I was using CentOS but I switched to Ubuntu to
>> try to build against some newer libraries.  I thought the problem was
>> resolved but it is very intermittent so its hard to tell.  The gdb
>> output is at http://pastebin.freeswitch.org/12451  The Freeswitch
>> process just seems to randomly start using 100% of the CPU and stop
>> responding about once every 12-36 hours.  Today it had to be  
>> restarted
>> twice, somedays not at all, very light load, about 100 calls per day.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Mar 24 22:43:25 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 00:43:25 -0500
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
Message-ID: <016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>

Odd nobody else has reported this behavior.  Have you viewed the wiki pages on this topic of running under a vm?
/b

On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:

> I dunno, just thought I'd try something different.. :-)  
> 
> root at sip:/usr/local/freeswitch/bin# uname -r
> 2.6.31-20-generic
> 
> I'd much rather use CentOS.  The Ubuntu machine is running under Xen 3.4.1 if that matters.
> 
> 

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From spencer at 5ninesolutions.com  Wed Mar 24 23:10:12 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 24 Mar 2010 23:10:12 -0700
Subject: [Freeswitch-users] Sofia crashing when registering
In-Reply-To: <016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
Message-ID: <03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>

Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes  
(1-n) -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is  
doing is bridging calls from Asterisk to the ITSP.  All media is  
routed point to point.  The dialplan is very simple as you can  
imagine.  If I disable registrations to Bandtel everything seems to be  
ok.  but to my eye I don't see anything weird about the format of the  
register that might cause a problem.

Spencer

On Mar 24, 2010, at 10:43 PM, Brian West wrote:

> Odd nobody else has reported this behavior.  Have you viewed the  
> wiki pages on this topic of running under a vm?
> /b
>
> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>
>> I dunno, just thought I'd try something different.. :-)
>>
>> root at sip:/usr/local/freeswitch/bin# uname -r
>> 2.6.31-20-generic
>>
>> I'd much rather use CentOS.  The Ubuntu machine is running under  
>> Xen 3.4.1 if that matters.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From thangappan143 at gmail.com  Wed Mar 24 21:40:02 2010
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 25 Mar 2010 10:10:02 +0530
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <191c3a031003240844p20d4cfadrd0c907b7a6eb6bf9@mail.gmail.com>
References: <7aa29e791003240133y4dc95a77r69a26f12b9b2f8f6@mail.gmail.com>
	<191c3a031003240844p20d4cfadrd0c907b7a6eb6bf9@mail.gmail.com>
Message-ID: <7aa29e791003242140y329f5ffele48ee2522dd185cd@mail.gmail.com>

When I done the post it has not shown in the FreeSWITCH-users-Archives.
Thats why another team member posted the same doubt. By usually when the
post is being done it is shown in the archives.







On Wed, Mar 24, 2010 at 9:14 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Did you not already send this message with another ID?
>
> You need the latest beta wanpipe driver and you need only 1 email address
> on this list.
>
>
>
> On Wed, Mar 24, 2010 at 3:33 AM, Thangappan.M <thangappan143 at gmail.com>wrote:
>
>> Dear all,
>>
>>     Compiled the wanpipe with freeswitch and configured it.
>>     When I originate a call from the freeswitch command line it said the
>> following error and got core dump.
>>
>>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753
>> &park()
>> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost
>> Version 103  Expecting 100
>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
>> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>>
>> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>>               wanpipe version is 3.5.8.6.
>>               smg_prid version is 1.63
>>
>> Please help me.........
>>
>> --
>> Regards,
>> Thangappan.M
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Thangappan.M
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From elihayun at gmail.com  Wed Mar 24 23:29:32 2010
From: elihayun at gmail.com (eliimap)
Date: Thu, 25 Mar 2010 08:29:32 +0200
Subject: [Freeswitch-users] Variable "default_language" has no effect on
	javascript
Message-ID: <4BAB02CC.8040602@gmail.com>

Hi

I set the variable "default_language" to be "ru" and then I run a js to
play a file


....

session.streamFile("conference/ronf-kicked.wav", "");

....



I got an error message


2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File
[/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav]
[System error : No such file or directory.]


that indicates that mod_sndfile is not checking this value but always
using en


Please check


Eli



From lists at infosecurity.ch  Thu Mar 25 00:42:15 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 25 Mar 2010 08:42:15 +0100
Subject: [Freeswitch-users] Move config between 16272M and 17084M does not
	work
Message-ID: <4BAB13D7.6050408@infosecurity.ch>

Hi,

yesterday i spent half a day trying to figure out what was happening in 
trying to move a configuration (that lookup dynamically via mod_xml_curl 
directory and dialplan) from FreeSWITCH version 16272M to 17084M .

I've done a clean install and just moved only the few lines required by 
hand.
Didn't worked:
     - On registration dynamic lookup works fine and users can login 
with password stored on remote server
     - On dial dynamic lookup of dialplan does not work and die with:
         2010-03-25 08:33:47.777741 [ERR] mod_dialplan_xml.c:440 Open of 
dialplan failed

I also tried to get the whoole /usr/local/freeswitch/conf of version 
"FreeSWITCH Version 1.0.5-20100109-0400 (16272M)" and put it as a 
replacement on top of the "FreeSWITCH Version 1.0.5-20100321-0400 
(17084M)" but the problem remain exactly the same:
     - Registration works via dynamic lookup via mod_xml_curl
     - Dialing does not work and dynamic lookup via mod_xml_curl does 
not work

In both case the configuration regarding mod_xml_curl is the following 
that works on 16272M but does not work on 17084M and is written in 
autoload_configs/xml_curl.conf.xml
<binding name="directory">
<param name="gateway-url" value="http://xxx/directory" 
bindings="directory"/>
<param name="auth-scheme" value="digest"/>
</binding>
<binding name="dialplan">
<param name="gateway-url" value="http://xxx/dialplan" bindings="dialplan"/>
</binding>

In both case i see in the log the correct initialization of xml_curl:
2010-03-25 08:29:50.193915 [NOTICE] mod_xml_curl.c:684 Binding 
[directory] XML Fetch Function [http://xxx/directory] [directory]
2010-03-25 08:29:50.193953 [NOTICE] mod_xml_curl.c:684 Binding 
[dialplan] XML Fetch Function [http://xxx/dialplan] [dialplan]
2010-03-25 08:29:50.196983 [CONSOLE] switch_loadable_module.c:900 
Successfully Loaded [mod_xml_curl]
2010-03-25 08:29:50.197019 [NOTICE] switch_loadable_module.c:272 Adding 
API Function 'xml_curl'

The same configuration works on 16272M but is broken on 17084M .

There's some modification that was done to dialplan and/or to 
mod_xml_curl that could cause this issue?

Is there some specific debugging i can do to understand if it's a bug 
(like it seems) or some configuration parameters changed regarding the 
dialplan lookup via mod_xml_curl ?

Waiting for yours

Fabio Pietrosanti



From rupa at rupa.com  Thu Mar 25 04:13:01 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 25 Mar 2010 06:13:01 -0500
Subject: [Freeswitch-users] Fwd:  Sofia crashing when registering
In-Reply-To: <03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
	<03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
Message-ID: <b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>

Next time it happens, prior to a restart of the freeswitch process, run the
fscore_pb script to get us a set of backtraces of the running process.

support-d/fscore_pb gcore <yourname>

try running it before hand to ensure you get expected results.  Getting a
core does pause the process for a short period of time, so do your test when
someone won't notice a RTP hickup.

On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes (1-n)
> -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is doing is
> bridging calls from Asterisk to the ITSP.  All media is routed point to
> point.  The dialplan is very simple as you can imagine.  If I disable
> registrations to Bandtel everything seems to be ok.  but to my eye I don't
> see anything weird about the format of the register that might cause a
> problem.
>
> Spencer
>
> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>
> Odd nobody else has reported this behavior.  Have you viewed the wiki pages
> on this topic of running under a vm?
> /b
>
> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>
> I dunno, just thought I'd try something different.. :-)
>
> root at sip:/usr/local/freeswitch/bin# uname -r
> 2.6.31-20-generic
>
> I'd much rather use CentOS.  The Ubuntu machine is running under Xen 3.4.1
> if that matters.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From lists at infosecurity.ch  Thu Mar 25 04:56:05 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 25 Mar 2010 12:56:05 +0100
Subject: [Freeswitch-users] Move config between 16272M and 17084M does
	not work
In-Reply-To: <4BAB13D7.6050408@infosecurity.ch>
References: <4BAB13D7.6050408@infosecurity.ch>
Message-ID: <4BAB4F55.6010506@infosecurity.ch>

Hi all,

i modified the source code in mod_dialplan_xml.c to distinguish between 
"Open of dialplan failed" error messages.

The call die on the dialplan_xml_locate that does not find a dialplan 
even if it's defined in xml_curl.conf.xml as described below and from 
the freeswitch.xml the loading of dialplan from file has been removed .

I noticed that FS 17084 vs 16272 have a lot of change in 
mod_dialplan_xml.c , maybe there's some bug and/or some new syntax to be 
used to make dynamic lookup of dialplan via xml_curl ?

Fabio

On 25/03/10 08.42, Fabio Pietrosanti (naif) wrote:
> Hi,
>
> yesterday i spent half a day trying to figure out what was happening 
> in trying to move a configuration (that lookup dynamically via 
> mod_xml_curl directory and dialplan) from FreeSWITCH version 16272M to 
> 17084M .
>
> I've done a clean install and just moved only the few lines required 
> by hand.
> Didn't worked:
>     - On registration dynamic lookup works fine and users can login 
> with password stored on remote server
>     - On dial dynamic lookup of dialplan does not work and die with:
>         2010-03-25 08:33:47.777741 [ERR] mod_dialplan_xml.c:440 Open 
> of dialplan failed
>
> I also tried to get the whoole /usr/local/freeswitch/conf of version 
> "FreeSWITCH Version 1.0.5-20100109-0400 (16272M)" and put it as a 
> replacement on top of the "FreeSWITCH Version 1.0.5-20100321-0400 
> (17084M)" but the problem remain exactly the same:
>     - Registration works via dynamic lookup via mod_xml_curl
>     - Dialing does not work and dynamic lookup via mod_xml_curl does 
> not work
>
> In both case the configuration regarding mod_xml_curl is the following 
> that works on 16272M but does not work on 17084M and is written in 
> autoload_configs/xml_curl.conf.xml
> <binding name="directory">
> <param name="gateway-url" value="http://xxx/directory" 
> bindings="directory"/>
> <param name="auth-scheme" value="digest"/>
> </binding>
> <binding name="dialplan">
> <param name="gateway-url" value="http://xxx/dialplan" 
> bindings="dialplan"/>
> </binding>
>
> In both case i see in the log the correct initialization of xml_curl:
> 2010-03-25 08:29:50.193915 [NOTICE] mod_xml_curl.c:684 Binding 
> [directory] XML Fetch Function [http://xxx/directory] [directory]
> 2010-03-25 08:29:50.193953 [NOTICE] mod_xml_curl.c:684 Binding 
> [dialplan] XML Fetch Function [http://xxx/dialplan] [dialplan]
> 2010-03-25 08:29:50.196983 [CONSOLE] switch_loadable_module.c:900 
> Successfully Loaded [mod_xml_curl]
> 2010-03-25 08:29:50.197019 [NOTICE] switch_loadable_module.c:272 
> Adding API Function 'xml_curl'
>
> The same configuration works on 16272M but is broken on 17084M .
>
> There's some modification that was done to dialplan and/or to 
> mod_xml_curl that could cause this issue?
>
> Is there some specific debugging i can do to understand if it's a bug 
> (like it seems) or some configuration parameters changed regarding the 
> dialplan lookup via mod_xml_curl ?
>
> Waiting for yours
>
> Fabio Pietrosanti
>



From brian at freeswitch.org  Thu Mar 25 06:26:26 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 08:26:26 -0500
Subject: [Freeswitch-users] Move config between 16272M and 17084M does
	not work
In-Reply-To: <4BAB13D7.6050408@infosecurity.ch>
References: <4BAB13D7.6050408@infosecurity.ch>
Message-ID: <A2E9CD4A-B3EB-44BD-8D1E-FE67B38416A2@freeswitch.org>

Nothing between these two should have broken... in fact configs from two years ago should work.  We work hard to make sure we don't break anything.

/b

On Mar 25, 2010, at 2:42 AM, Fabio Pietrosanti (naif) wrote:

> The same configuration works on 16272M but is broken on 17084M .



From anthony.minessale at gmail.com  Thu Mar 25 06:26:58 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 08:26:58 -0500
Subject: [Freeswitch-users] Fwd: Sofia crashing when registering
In-Reply-To: <b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<20100325044200.GA7885@jdc.jasonjgw.net>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
	<03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
	<b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>
Message-ID: <191c3a031003250626h74092581g14aa4185e69d02ca@mail.gmail.com>

If you recall, I tried to help you with this already once.
I spent a long time on your box building out fresh trunk and configuring it
to trace the problem.
Then you told me you erased it all by accident =/

I suppose we can try again but I would appreciate if if you gather the info
for me.

edit conf/autoload_configs/logfile.conf.xml

look for this line:

<map name="all" value="debug,info,notice,warning,err,crit,alert"/>

replace it with:

<map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>


in the cli:

sofia profile internal siptrace on
sofia loglevel all 9

(repeat for any other profiles)

also run wireshark or tcpdump to capture all traffic to that particular
provider

When it happens, stop FS and the capture and get me access to the resulting
files.

This is a rare issue triggered by some strange sip message.
What we are trying to do here is get a copy of the traffic that triggers it
so we can stop it.



On Thu, Mar 25, 2010 at 6:13 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> Next time it happens, prior to a restart of the freeswitch process, run the
> fscore_pb script to get us a set of backtraces of the running process.
>
> support-d/fscore_pb gcore <yourname>
>
> try running it before hand to ensure you get expected results.  Getting a
> core does pause the process for a short period of time, so do your test when
> someone won't notice a RTP hickup.
>
> On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <
> spencer at 5ninesolutions.com> wrote:
>
>> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes (1-n)
>> -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is doing is
>> bridging calls from Asterisk to the ITSP.  All media is routed point to
>> point.  The dialplan is very simple as you can imagine.  If I disable
>> registrations to Bandtel everything seems to be ok.  but to my eye I don't
>> see anything weird about the format of the register that might cause a
>> problem.
>>
>> Spencer
>>
>> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>>
>> Odd nobody else has reported this behavior.  Have you viewed the wiki
>> pages on this topic of running under a vm?
>> /b
>>
>> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>>
>> I dunno, just thought I'd try something different.. :-)
>>
>> root at sip:/usr/local/freeswitch/bin# uname -r
>> 2.6.31-20-generic
>>
>> I'd much rather use CentOS.  The Ubuntu machine is running under Xen 3.4.1
>> if that matters.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 25 06:27:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 08:27:59 -0500
Subject: [Freeswitch-users] Fwd: Sofia crashing when registering
In-Reply-To: <191c3a031003250626h74092581g14aa4185e69d02ca@mail.gmail.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
	<03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
	<b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>
	<191c3a031003250626h74092581g14aa4185e69d02ca@mail.gmail.com>
Message-ID: <191c3a031003250627r71d5a731p447b207f51fe96eb@mail.gmail.com>

since I forgot to mention,

restart FS or at leasr mod_logfile after making the changes to the config


On Thu, Mar 25, 2010 at 8:26 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If you recall, I tried to help you with this already once.
> I spent a long time on your box building out fresh trunk and configuring it
> to trace the problem.
> Then you told me you erased it all by accident =/
>
> I suppose we can try again but I would appreciate if if you gather the info
> for me.
>
> edit conf/autoload_configs/logfile.conf.xml
>
> look for this line:
>
> <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
> replace it with:
>
> <map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>
>
>
> in the cli:
>
> sofia profile internal siptrace on
> sofia loglevel all 9
>
> (repeat for any other profiles)
>
> also run wireshark or tcpdump to capture all traffic to that particular
> provider
>
> When it happens, stop FS and the capture and get me access to the resulting
> files.
>
> This is a rare issue triggered by some strange sip message.
> What we are trying to do here is get a copy of the traffic that triggers it
> so we can stop it.
>
>
>
>
> On Thu, Mar 25, 2010 at 6:13 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> Next time it happens, prior to a restart of the freeswitch process, run
>> the fscore_pb script to get us a set of backtraces of the running process.
>>
>> support-d/fscore_pb gcore <yourname>
>>
>> try running it before hand to ensure you get expected results.  Getting a
>> core does pause the process for a short period of time, so do your test when
>> someone won't notice a RTP hickup.
>>
>> On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <
>> spencer at 5ninesolutions.com> wrote:
>>
>>> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes
>>> (1-n) -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is doing is
>>> bridging calls from Asterisk to the ITSP.  All media is routed point to
>>> point.  The dialplan is very simple as you can imagine.  If I disable
>>> registrations to Bandtel everything seems to be ok.  but to my eye I don't
>>> see anything weird about the format of the register that might cause a
>>> problem.
>>>
>>> Spencer
>>>
>>> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>>>
>>> Odd nobody else has reported this behavior.  Have you viewed the wiki
>>> pages on this topic of running under a vm?
>>> /b
>>>
>>> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>>>
>>> I dunno, just thought I'd try something different.. :-)
>>>
>>> root at sip:/usr/local/freeswitch/bin# uname -r
>>> 2.6.31-20-generic
>>>
>>> I'd much rather use CentOS.  The Ubuntu machine is running under Xen
>>> 3.4.1 if that matters.
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Thu Mar 25 06:30:43 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 08:30:43 -0500
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>
References: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
	<191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>
	<1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>
Message-ID: <191c3a031003250630v17eb7d6fv94aa29971352ddea@mail.gmail.com>

you need to use the beta not the one that is suggested.


On Wed, Mar 24, 2010 at 11:52 PM, velusamy Krishnan <
velu.technical at gmail.com> wrote:

> Installed the wanpipe version(wanpipe-3.5.10.10b.tgz) which is suggested in
> the http://wiki.sangoma.com/wanpipe-freeswitch-install.
>
> But, the same is happening.
> >From sangoma_mgd.log file found the following,
>
> sangoma_prid: = Sangoma PRI Protocol Stack Daemon
> sangoma_prid: = Version: 1.75
> sangoma_prid: = Date: Mar 18 2010
> sangoma_prid: = Wanpipe Release: wanpipe-3.5.10.10
> sangoma_prid: = Revision:Revision: 15921
>
> Do you mean some other wanpipe release?
>
>
> On Wed, Mar 24, 2010 at 8:28 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you need the newest beta revision of wanpipe to use with latest trunk
>>
>> On Wed, Mar 24, 2010 at 6:29 AM, velusamy Krishnan <
>> velu.technical at gmail.com> wrote:
>>
>>> Dear all,
>>>
>>>     Compiled the wanpipe with freeswitch and configured it.
>>>     When I originate a call from the freeswitch command line it said the
>>> following error and got core dump.
>>>
>>>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753
>>> &park()
>>> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid
>>> Boost Version 103  Expecting 100
>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
>>> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>>>
>>> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>>>               wanpipe version is 3.5.8.6.
>>>               smg_prid version is 1.63
>>>
>>> Please help me.........
>>>
>>> Thanks..
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Thu Mar 25 06:40:15 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Thu, 25 Mar 2010 14:40:15 +0100
Subject: [Freeswitch-users] Move config between 16272M and 17084M does
 not work
In-Reply-To: <A2E9CD4A-B3EB-44BD-8D1E-FE67B38416A2@freeswitch.org>
References: <4BAB13D7.6050408@infosecurity.ch>
	<A2E9CD4A-B3EB-44BD-8D1E-FE67B38416A2@freeswitch.org>
Message-ID: <4BAB67BF.2080005@infosecurity.ch>

I followed on both installation the same procedure:
mkdir 23.03.2010-FS
cd 23.03.2010-FS
apt-get install subversion autoconf automake libtool gcc g++ make 
ncurses-dev libexpat1-dev  zlib1g-dev libssl-dev libgnutls-dev 
libcurl3-openssl-dev automake1.9

wget http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz
tar zxvf freeswitch-1.0.5-latest.tar.gz
cd freeswitch-1.0.5-20100321-0400
vim modules.conf
     xml_int/mod_xml_curl
     applications/mod_curl
     applications/mod_cidlookup
     dialplans/mod_dialplan_directory
     applications/mod_directory
     directories/mod_ldap
     endpoints/mod_dingaling
     say/mod_say_it
     applications/mod_spy
     applications/mod_stress

# 23.03.2010
make current #update from SVN
make samples
make install && make cd-sounds-install && make cd-moh-install

Later i just made:
mv /usr/local/freeswitch/conf /usr/local/freeswitch/conf.disabled
cp -pr /usr/local/freeswitch.old/conf /usr/local/freeswitch/conf

So the configurations are fully aligned (is the same), the modules.conf 
during the compilation is aligned, the operating system is aligned 
(debian/stable up-to-date), verified that mod\* is aligned .

The dynamic xml_curl lookup for REGISTER works.
The dynamic xml_curl lookup for dialplan does not work.

The function that fail is dialplan_xml_locate with error:
2010-03-25 11:59:42.877926 [ERR] mod_dialplan_xml.c:440 Open of dialplan 
failed

I verified with tcpdump that it does not even try to make the xml_curl 
lookup .

The configuration file of xml_curl.conf.xml is *the same* .

I noticed that mod_dialplan_xml.c changed in several parts between 16272 
and 17084 revisions.

Some place where to look at for a deep investigation?
Enable dialplan debugging somehow?

Fabio
On 25/03/10 14.26, Brian West wrote:
> Nothing between these two should have broken... in fact configs from two years ago should work.  We work hard to make sure we don't break anything.
>
> /b
>
> On Mar 25, 2010, at 2:42 AM, Fabio Pietrosanti (naif) wrote:
>
>    
>> The same configuration works on 16272M but is broken on 17084M .
>>      
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    



From moises.silva at gmail.com  Thu Mar 25 06:58:42 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 25 Mar 2010 09:58:42 -0400
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>
References: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
	<191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>
	<1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>
Message-ID: <c4d05cbe1003250658ue743dd4yfd4be2062884c1d7@mail.gmail.com>

Actually the log message indicates you have older openzap and newer wanpipe
(OpenZAP expects 100 and is receiving 103, therefore OpenZAP is the outdated
here).

Update to latest openzap.

On Thu, Mar 25, 2010 at 12:52 AM, velusamy Krishnan <
velu.technical at gmail.com> wrote:

> Installed the wanpipe version(wanpipe-3.5.10.10b.tgz) which is suggested in
> the http://wiki.sangoma.com/wanpipe-freeswitch-install.
>
> But, the same is happening.
> >From sangoma_mgd.log file found the following,
>
> sangoma_prid: = Sangoma PRI Protocol Stack Daemon
> sangoma_prid: = Version: 1.75
> sangoma_prid: = Date: Mar 18 2010
> sangoma_prid: = Wanpipe Release: wanpipe-3.5.10.10
> sangoma_prid: = Revision:Revision: 15921
>
> Do you mean some other wanpipe release?
>
> On Wed, Mar 24, 2010 at 8:28 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you need the newest beta revision of wanpipe to use with latest trunk
>>
>> On Wed, Mar 24, 2010 at 6:29 AM, velusamy Krishnan <
>> velu.technical at gmail.com> wrote:
>>
>>> Dear all,
>>>
>>>     Compiled the wanpipe with freeswitch and configured it.
>>>     When I originate a call from the freeswitch command line it said the
>>> following error and got core dump.
>>>
>>>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753
>>> &park()
>>> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid
>>> Boost Version 103  Expecting 100
>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
>>> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>>>
>>> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>>>               wanpipe version is 3.5.8.6.
>>>               smg_prid version is 1.63
>>>
>>> Please help me.........
>>>
>>> Thanks..
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From lloyd.aloysius at gmail.com  Thu Mar 25 07:14:49 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 25 Mar 2010 10:14:49 -0400
Subject: [Freeswitch-users] How to force a Extension use g729 only
Message-ID: <8a19bf2e1003250714j4c105174sb9386ed4609fa7dd@mail.gmail.com>

Hi All,

I have the following setup

[IP Phones] ----[Router] --------------[FreeSWITCH] ------ [ITSP]

How to force a extension use g729 only. I could not find any parameter to
set for each extension.

In Asterisk world I can do

disallow=all
allow=g729


Thanks,
Lloyd
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From kenfulmer at icstechnologysolutions.com  Thu Mar 25 07:21:16 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 25 Mar 2010 09:21:16 -0500
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <20100324220027.722BF31B5F8@cuneorg-email.cune.pri>
References: <00c901cacb94$e9e92710$bdbb7530$@com>
	<20100324220027.722BF31B5F8@cuneorg-email.cune.pri>
Message-ID: <00b801cacc26$6e9ae930$4bd0bb90$@com>

There's a file called freeswitch.pid in the path:  opt/freeswitch/run/
freeswitch.pid

I changed the permissions for the file to read / write / execute. However, I
still received the same error. I rebooted the Ubuntu server and the
permissions reverted back to the previous setting. 

Thanks,

Ken

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Russell.Mosemann at cune.org
Sent: Wednesday, March 24, 2010 5:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot lock pid

Ken Fulmer <kenfulmer at icstechnologysolutions.com> said:

> We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
> server. Has anyone seen this before?

It looks like a permission problem. Can freeswitch access the path to

/what/ever/freeswitch/log/freeswitch.pid

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Mar 25 07:21:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 09:21:45 -0500
Subject: [Freeswitch-users] Move config between 16272M and 17084M does
	not work
In-Reply-To: <4BAB67BF.2080005@infosecurity.ch>
References: <4BAB13D7.6050408@infosecurity.ch>
	<A2E9CD4A-B3EB-44BD-8D1E-FE67B38416A2@freeswitch.org>
	<4BAB67BF.2080005@infosecurity.ch>
Message-ID: <191c3a031003250721i15d83221l56e007d46d1a9c80@mail.gmail.com>

Tracked down to a patch that caused a regression in r16997
update to latest and the patch is reverted.


On Thu, Mar 25, 2010 at 8:40 AM, Fabio Pietrosanti (naif) <
lists at infosecurity.ch> wrote:

> I followed on both installation the same procedure:
> mkdir 23.03.2010-FS
> cd 23.03.2010-FS
> apt-get install subversion autoconf automake libtool gcc g++ make
> ncurses-dev libexpat1-dev  zlib1g-dev libssl-dev libgnutls-dev
> libcurl3-openssl-dev automake1.9
>
> wget http://latest.freeswitch.org/freeswitch-1.0.5-latest.tar.gz
> tar zxvf freeswitch-1.0.5-latest.tar.gz
> cd freeswitch-1.0.5-20100321-0400
> vim modules.conf
>     xml_int/mod_xml_curl
>     applications/mod_curl
>     applications/mod_cidlookup
>     dialplans/mod_dialplan_directory
>     applications/mod_directory
>     directories/mod_ldap
>     endpoints/mod_dingaling
>     say/mod_say_it
>     applications/mod_spy
>     applications/mod_stress
>
> # 23.03.2010
> make current #update from SVN
> make samples
> make install && make cd-sounds-install && make cd-moh-install
>
> Later i just made:
> mv /usr/local/freeswitch/conf /usr/local/freeswitch/conf.disabled
> cp -pr /usr/local/freeswitch.old/conf /usr/local/freeswitch/conf
>
> So the configurations are fully aligned (is the same), the modules.conf
> during the compilation is aligned, the operating system is aligned
> (debian/stable up-to-date), verified that mod\* is aligned .
>
> The dynamic xml_curl lookup for REGISTER works.
> The dynamic xml_curl lookup for dialplan does not work.
>
> The function that fail is dialplan_xml_locate with error:
> 2010-03-25 11:59:42.877926 [ERR] mod_dialplan_xml.c:440 Open of dialplan
> failed
>
> I verified with tcpdump that it does not even try to make the xml_curl
> lookup .
>
> The configuration file of xml_curl.conf.xml is *the same* .
>
> I noticed that mod_dialplan_xml.c changed in several parts between 16272
> and 17084 revisions.
>
> Some place where to look at for a deep investigation?
> Enable dialplan debugging somehow?
>
> Fabio
> On 25/03/10 14.26, Brian West wrote:
> > Nothing between these two should have broken... in fact configs from two
> years ago should work.  We work hard to make sure we don't break anything.
> >
> > /b
> >
> > On Mar 25, 2010, at 2:42 AM, Fabio Pietrosanti (naif) wrote:
> >
> >
> >> The same configuration works on 16272M but is broken on 17084M .
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From testeador01 at gmail.com  Thu Mar 25 07:32:23 2010
From: testeador01 at gmail.com (Milena)
Date: Thu, 25 Mar 2010 09:32:23 -0500
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <00b801cacc26$6e9ae930$4bd0bb90$@com>
References: <00c901cacb94$e9e92710$bdbb7530$@com>
	<20100324220027.722BF31B5F8@cuneorg-email.cune.pri>
	<00b801cacc26$6e9ae930$4bd0bb90$@com>
Message-ID: <c29b930b1003250732t5644d3dbi60177dd919dd69e4@mail.gmail.com>

Are you sure it isn't started yet?:
try to open* freeswitch/bin/fs_cli* instead of *freeswitch/bin/freeswitch*


2010/3/25 Ken Fulmer <kenfulmer at icstechnologysolutions.com>

> There's a file called freeswitch.pid in the path:  opt/freeswitch/run/
> freeswitch.pid
>
> I changed the permissions for the file to read / write / execute. However,
> I
> still received the same error. I rebooted the Ubuntu server and the
> permissions reverted back to the previous setting.
>
> Thanks,
>
> Ken
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Russell.Mosemann at cune.org
> Sent: Wednesday, March 24, 2010 5:00 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Cannot lock pid
>
> Ken Fulmer <kenfulmer at icstechnologysolutions.com> said:
>
> > We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
> > server. Has anyone seen this before?
>
> It looks like a permission problem. Can freeswitch access the path to
>
> /what/ever/freeswitch/log/freeswitch.pid
>
> --
> Russell Mosemann
>
>
>
> ________________________________________________________
> Concordia University, Nebraska
> See http://www.cune.edu/ for the latest news and events!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kenfulmer at icstechnologysolutions.com  Thu Mar 25 08:04:18 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 25 Mar 2010 10:04:18 -0500
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <00b801cacc26$6e9ae930$4bd0bb90$@com>
References: <00c901cacb94$e9e92710$bdbb7530$@com>	<20100324220027.722BF31B5F8@cuneorg-email.cune.pri>
	<00b801cacc26$6e9ae930$4bd0bb90$@com>
Message-ID: <00ce01cacc2c$7334d4f0$599e7ed0$@com>

I realized I wasn't entering the command properly. Sorry for the confusion.

However, I'm now getting the following error:

-bash: /opt/freeswitch/bin/freeswitch: Text file busy

Does anyone know why we're seeing that error?

Thanks,
Ken


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Thursday, March 25, 2010 9:21 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot lock pid

There's a file called freeswitch.pid in the path:  opt/freeswitch/run/
freeswitch.pid

I changed the permissions for the file to read / write / execute. However, I
still received the same error. I rebooted the Ubuntu server and the
permissions reverted back to the previous setting. 

Thanks,

Ken

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Russell.Mosemann at cune.org
Sent: Wednesday, March 24, 2010 5:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot lock pid

Ken Fulmer <kenfulmer at icstechnologysolutions.com> said:

> We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
> server. Has anyone seen this before?

It looks like a permission problem. Can freeswitch access the path to

/what/ever/freeswitch/log/freeswitch.pid

-- 
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From brian at freeswitch.org  Thu Mar 25 08:11:34 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 10:11:34 -0500
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <00ce01cacc2c$7334d4f0$599e7ed0$@com>
References: <00c901cacb94$e9e92710$bdbb7530$@com>	<20100324220027.722BF31B5F8@cuneorg-email.cune.pri>
	<00b801cacc26$6e9ae930$4bd0bb90$@com>
	<00ce01cacc2c$7334d4f0$599e7ed0$@com>
Message-ID: <BE121B9D-4046-4238-97B3-CF57C385C07F@freeswitch.org>

Because its running 

/b

On Mar 25, 2010, at 10:04 AM, Ken Fulmer wrote:

> I realized I wasn't entering the command properly. Sorry for the confusion.
> 
> However, I'm now getting the following error:
> 
> -bash: /opt/freeswitch/bin/freeswitch: Text file busy
> 
> Does anyone know why we're seeing that error?
> 
> Thanks,
> Ken



From mranga at gmail.com  Thu Mar 25 08:13:04 2010
From: mranga at gmail.com (M. Ranganathan)
Date: Thu, 25 Mar 2010 11:13:04 -0400
Subject: [Freeswitch-users] B2BUA behavior.
Message-ID: <5c7eebdd1003250813k661b9cf0mb5ac3f23dc11a398@mail.gmail.com>

Hello,

I am learning about FreeSWITCH.

If I allocate a B2BUA in freeSWITCH and send it a REFER, will the
REFER be passed through transparently or converted to an INVITE? If it
is the latter behavior, is it possible to configure FreeSWITCH to pass
the REFER through?

Thanks


-- 
M. Ranganathan


From kenfulmer at icstechnologysolutions.com  Thu Mar 25 08:25:22 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 25 Mar 2010 10:25:22 -0500
Subject: [Freeswitch-users] Cannot lock pid
In-Reply-To: <c29b930b1003250732t5644d3dbi60177dd919dd69e4@mail.gmail.com>
References: <00c901cacb94$e9e92710$bdbb7530$@com>	<20100324220027.722BF31B5F8@cuneorg-email.cune.pri>	<00b801cacc26$6e9ae930$4bd0bb90$@com>
	<c29b930b1003250732t5644d3dbi60177dd919dd69e4@mail.gmail.com>
Message-ID: <00db01cacc2f$62c30e40$28492ac0$@com>

Sorry guys. I was connected via SSH. I opened fs_cli and everything seems to
be working properly. Thanks for your patience!

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena
Sent: Thursday, March 25, 2010 9:32 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot lock pid

 

 

Are you sure it isn't started yet?:

try to open freeswitch/bin/fs_cli instead of freeswitch/bin/freeswitch

 

 

2010/3/25 Ken Fulmer <kenfulmer at icstechnologysolutions.com>

There's a file called freeswitch.pid in the path:  opt/freeswitch/run/
freeswitch.pid

I changed the permissions for the file to read / write / execute. However, I
still received the same error. I rebooted the Ubuntu server and the
permissions reverted back to the previous setting.

Thanks,

Ken


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Russell.Mosemann at cune.org
Sent: Wednesday, March 24, 2010 5:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot lock pid

Ken Fulmer <kenfulmer at icstechnologysolutions.com> said:

> We see the above error when attempting to start FreeSwitch on Ubuntu 9.10
> server. Has anyone seen this before?

It looks like a permission problem. Can freeswitch access the path to

/what/ever/freeswitch/log/freeswitch.pid

--
Russell Mosemann



________________________________________________________
Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From brian at freeswitch.org  Thu Mar 25 08:28:29 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 10:28:29 -0500
Subject: [Freeswitch-users] B2BUA behavior.
In-Reply-To: <5c7eebdd1003250813k661b9cf0mb5ac3f23dc11a398@mail.gmail.com>
References: <5c7eebdd1003250813k661b9cf0mb5ac3f23dc11a398@mail.gmail.com>
Message-ID: <9F02AAF5-590B-45CB-9E8C-3E7185B53C28@freeswitch.org>

Sounds like you need to be using a Proxy instead of a B2BUA

/b

On Mar 25, 2010, at 10:13 AM, M. Ranganathan wrote:

> Hello,
> 
> I am learning about FreeSWITCH.
> 
> If I allocate a B2BUA in freeSWITCH and send it a REFER, will the
> REFER be passed through transparently or converted to an INVITE? If it
> is the latter behavior, is it possible to configure FreeSWITCH to pass
> the REFER through?



From williams.owen at gmail.com  Thu Mar 25 08:51:30 2010
From: williams.owen at gmail.com (OwenW)
Date: Thu, 25 Mar 2010 07:51:30 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <87pr6yz5wa.fsf@quasar.esben-stien.name>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
Message-ID: <1269532290668-4798438.post@n2.nabble.com>


I get the same message when trying to playback.

I've added a few debug lines - it looks as if the 'version' is never written
to the file during video record.

I believe this is because the following line (in mod_fsv) returns false:
if (switch_channel_test_flag(channel, CF_VIDEO)) {

Maybe I should infer from this that the channel I'm trying to record isn't
sending video.  I'm using XLite on windows - I start a 'video call' and
whilst the call is started I suspect it hasn't been able to negotiate video
and perhaps fallen back to audio.  XLite does H.263 and H.263+.

Owen
-- 
View this message in context: http://n2.nabble.com/Freeswitch-Video-Capture-and-Playback-tp4035796p4798438.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From wchao at yahoo.com  Thu Mar 25 09:25:23 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Thu, 25 Mar 2010 12:25:23 -0400 (EDT)
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
 phones
In-Reply-To: <8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>

Gabe,

My Linksys phones do seem to reboot every time they sync the config. 
However, I have the Message_Waiting option turned off. Is there another 
option that you think might be related to the rebooting? I do not think it 
is a DHCP issue because the phones get the same IP address each time they 
request one from the DHCP server. I am using the 5.1.8 firmware, which I 
believe is the latest for the 921 and 941 phones.

I have put one of my configuration files online here:
http://209.190.22.48/fs/linksys/

I changed a few things like username/password/proxy/provisioning URL, but 
otherwise left everything the same as one of my real configuration files.

Do you have a sample config you could send to me for a Linksys 901, 921, 
or 941? I would love to be able to compare to see what other options might 
make a difference.

Regards,
Wellie


Date: Wed, 24 Mar 2010 20:42:38 -0700
From: Gabriel Kuri <gkuri at ieee.org>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
    phones

If you're using central provisioning with the Linksys/Cisco Small
Business series phones and you leave the Message_Waiting option in the
config (which is the default, very stupid), the phones will reboot
every time they sync the config, if they have a voicemail waiting.

Check to see if the Linksys spontaneous reboots correlate with pulling
down the config (assuming you're using central provisioning) and make
sure you either comment or delete Message_Waiting from the config.

You may also want to see if there's any correlation with DHCP and the
phones renewing their IP. We put all of our phones on DHCP
reservations, so their IP doesn't/shouldn't change. I'm not sure what
the phone will do if it goes to renew its address and the DHCP server
hands it a different IP, it may cause the phone to reboot?

Cheers,
Gabe


On Wed, Mar 24, 2010 at 6:35 PM, Wellie Chao <wchao at yahoo.com> wrote:
> I am using Freeswitch and I have some phones that spontaneously reboot
> (often in the middle of a call).
>
> I did some searching around on the web, but didn't find any information
> that really applied to the problem I'm experiencing. I have a few dozen
> Polycom phones and a handful of Linksys phones of various models: Polycom
> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921,
> SPA941.
>
> I also have snom phones, but they don't reboot spontaneously. It's only
> the Polycom and Linksys phones that reboot spontaneously, and it seems to
> be correlated to call volume. Heavy usage of the Polycom and Linksys
> phones causes them to reboot more frequently (i.e. shorter duration
> between the reboots). For people making 50 calls a day, the Polycom phones
> spontaneously reboot every three days or so (and always on a call, it
> seems). For people making one or two calls a day, the Polycom will go
> months without a spontaneous reboot. This correlation between call volume
> and reboot frequency leads me to believe that there is some bug in the
> Polycom firmware, perhaps some of overflow condition that accumulates over
> time and eventually causes a crash (and subsequent reboot).
>
> I was wondering if anybody else experienced something similar, and whether
> anyone had developed a solution. One thing I am considering is to set up a
> script to reboot the Polycom phones in the middle of the night every day,
> which would hopefully prevent them from spontaneously rebooting during the
> day by preventing the phones from ever reaching the overflow condition (if
> that is indeed what causes the reboots). A nightly reboot would be a
> hackish way of fixing the problem, so I'd also love to hear if anybody has
> any other speculations on what might be causing the spontaneous reboots.
> We have phones in multiple buildings, and we've discovered that the
> spontaneous reboots happen on phones on different networks, so it's not
> specific to a particular LAN segment. For the most part, we are using the
> standard config files for the Polycom phones and for the Linksys phones.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From kenfulmer at icstechnologysolutions.com  Thu Mar 25 09:25:14 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Thu, 25 Mar 2010 11:25:14 -0500
Subject: [Freeswitch-users] Confusion - Please Help
Message-ID: <00fc01cacc37$bfc05140$3f40f3c0$@com>

We need the internal sipX box to register phones and send / receive PSTN
calls through the FS SBC.  I don't understand how we need to setup SIP
Profiles in FS. We basically need one gateway to an ITSP and one gateway to
our sipX PBX. Is this done in different places? 

 

Thanks,

 

Ken Fulmer

 

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From brian at freeswitch.org  Thu Mar 25 09:28:57 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 11:28:57 -0500
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
	phones
In-Reply-To: <alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
	<alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>
Message-ID: <71E19730-C831-4398-B042-BF8BFC4BBB04@freeswitch.org>

Do you generate your configs on request?

/b

On Mar 25, 2010, at 11:25 AM, Wellie Chao wrote:

> Gabe,
> 
> My Linksys phones do seem to reboot every time they sync the config. 
> However, I have the Message_Waiting option turned off. Is there another 
> option that you think might be related to the rebooting? I do not think it 
> is a DHCP issue because the phones get the same IP address each time they 
> request one from the DHCP server. I am using the 5.1.8 firmware, which I 
> believe is the latest for the 921 and 941 phones.
> 
> I have put one of my configuration files online here:
> http://209.190.22.48/fs/linksys/
> 
> I changed a few things like username/password/proxy/provisioning URL, but 
> otherwise left everything the same as one of my real configuration files.
> 
> Do you have a sample config you could send to me for a Linksys 901, 921, 
> or 941? I would love to be able to compare to see what other options might 
> make a difference.
> 
> Regards,
> Wellie

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From infos at madovsky.org  Thu Mar 25 09:33:04 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 12:33:04 -0400
Subject: [Freeswitch-users] Confusion - Please Help
References: <00fc01cacc37$bfc05140$3f40f3c0$@com>
Message-ID: <05DA810FE3954B31A35273B9A91F7410@MOBILEE1705>

look at external and internal sip profiles also public and default dialplan
  ----- Original Message ----- 
  From: Ken Fulmer 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 12:25 PM
  Subject: [Freeswitch-users] Confusion - Please Help


  We need the internal sipX box to register phones and send / receive PSTN calls through the FS SBC.  I don't understand how we need to setup SIP Profiles in FS. We basically need one gateway to an ITSP and one gateway to our sipX PBX. Is this done in different places? 

   

  Thanks,

   

  Ken Fulmer

   



------------------------------------------------------------------------------


  _______________________________________________
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From msc at freeswitch.org  Thu Mar 25 09:33:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 09:33:31 -0700
Subject: [Freeswitch-users] IRC Day!
Message-ID: <87f2f3b91003250933g43da6e5rd275551e71c46f49@mail.gmail.com>

Hello all FreeSWITCHers!

We invite everyone around the world to join us in #freeswitch on
irc.freenode.net. We want everyone to be online today, even if you just
leave your IRC client running in the background. Also, if you don't have an
IRC client handy then go to http://www.freeswitch.org/ and use the web login
located on the right-hand side bar (below the features list). You'll need a
Java-enable browser.

Everyone have a great day!
-Michael
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From msc at freeswitch.org  Thu Mar 25 09:42:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 09:42:13 -0700
Subject: [Freeswitch-users] Variable "default_language" has no effect on
	javascript
In-Reply-To: <4BAB02CC.8040602@gmail.com>
References: <4BAB02CC.8040602@gmail.com>
Message-ID: <87f2f3b91003250942h77de93a6y4f8b9ee1767b7fd9@mail.gmail.com>

Ummm
You're looking for the file "ronf-kicked.wav" - I'm pretty sure that's not
what you want. Try "conf-kicked.wav" and see if that's any better....
-MC

On Wed, Mar 24, 2010 at 11:29 PM, eliimap <elihayun at gmail.com> wrote:

> Hi
>
> I set the variable "default_language" to be "ru" and then I run a js to
> play a file
>
>
> ....
>
> session.streamFile("conference/ronf-kicked.wav", "");
>
> ....
>
>
>
> I got an error message
>
>
> 2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File
>
> [/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav]
> [System error : No such file or directory.]
>
>
> that indicates that mod_sndfile is not checking this value but always
> using en
>
>
> Please check
>
>
> Eli
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Mar 25 09:47:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 09:47:02 -0700
Subject: [Freeswitch-users] How to force a Extension use g729 only
In-Reply-To: <8a19bf2e1003250714j4c105174sb9386ed4609fa7dd@mail.gmail.com>
References: <8a19bf2e1003250714j4c105174sb9386ed4609fa7dd@mail.gmail.com>
Message-ID: <87f2f3b91003250947u305a9840h3c1b142a3ec833f2@mail.gmail.com>

Try adding this to your directory entry for the user in the <variables>
node:

<variable name="absolute_codec_string" value="g729"/>

-MC

On Thu, Mar 25, 2010 at 7:14 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> I have the following setup
>
> [IP Phones] ----[Router] --------------[FreeSWITCH] ------ [ITSP]
>
> How to force a extension use g729 only. I could not find any parameter to
> set for each extension.
>
> In Asterisk world I can do
>
> disallow=all
> allow=g729
>
>
> Thanks,
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wchao at yahoo.com  Thu Mar 25 09:52:40 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Thu, 25 Mar 2010 12:52:40 -0400 (EDT)
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
 phones
In-Reply-To: <30663396-E364-47C9-95E8-F14BA773497E@freeswitch.org>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<30663396-E364-47C9-95E8-F14BA773497E@freeswitch.org>
Message-ID: <alpine.LRH.2.00.1003251237540.5523@osgood.microoffice.com>

Thanks for responding, Brian, Chris, and Ghulam. To answer your questions, 
I am using bootrom 4.2.1 or 4.1.4 (depending on the phone model because 
some don't support the later bootrom) and also SIP 3.2.2 or 3.1.4, again 
depending on the phone model because of lack of support for 3.2.2 for some 
of the older Polycom models. I'm using the split SIP image.

With regards to Chris's suggestion about switches, we have the Polycom 
phones hooked up to Linksys managed switches and Dell managed switches, 
and the Polycom phones spontaneously reboot on both switches. I'm pretty 
sure the Dell managed switches are not dropping link. I don't think the 
Linksys managed switches are either, but I'm less sure there because we've 
had some problems with the web management interface being a piece of crap 
in the past.

With regards to Ghulam's observation about speakerphone use, our Polycom 
phones reboot even when not on speakerphone, so that's not it.

I've uploaded some of my config files here:
http://209.190.22.48/fs/polycom/

They are the same as my real config files, except username, password, and 
hostnames. For the most part, they are the factory default settings.

I would love to see any config files you might be willing to share, to 
compare your config settings to mine.

Also, one thing that came to mind: how are you specifying the server and 
proxy addresses? I am just using a hard-coded IP address with no DNS 
magic. Do you think that would make a difference?

Regards,
Wellie


Date: Wed, 24 Mar 2010 20:44:55 -0500
From: Brian West <brian at freeswitch.org>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
    phones

I haven't... and I use the shit out of my polycom's, I use them for 10 hours or more solid a day on a single call.. while the 670 on my right makes 30-40 calls or more a day depending on what I'm doing.  Now you failed to tell us what Firmware Rev you're running and what options you have turned on?

/b

On Mar 24, 2010, at 8:35 PM, Wellie Chao wrote:

> I am using Freeswitch and I have some phones that spontaneously reboot 
> (often in the middle of a call).
> 
> I did some searching around on the web, but didn't find any information 
> that really applied to the problem I'm experiencing. I have a few dozen 
> Polycom phones and a handful of Linksys phones of various models: Polycom 
> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921, 
> SPA941.
> 
> I also have snom phones, but they don't reboot spontaneously. It's only 
> the Polycom and Linksys phones that reboot spontaneously, and it seems to 
> be correlated to call volume. Heavy usage of the Polycom and Linksys 
> phones causes them to reboot more frequently (i.e. shorter duration 
> between the reboots). For people making 50 calls a day, the Polycom phones 
> spontaneously reboot every three days or so (and always on a call, it 
> seems). For people making one or two calls a day, the Polycom will go 
> months without a spontaneous reboot. This correlation between call volume 
> and reboot frequency leads me to believe that there is some bug in the 
> Polycom firmware, perhaps some of overflow condition that accumulates over 
> time and eventually causes a crash (and subsequent reboot).
> 
> I was wondering if anybody else experienced something similar, and whether 
> anyone had developed a solution. One thing I am considering is to set up a 
> script to reboot the Polycom phones in the middle of the night every day, 
> which would hopefully prevent them from spontaneously rebooting during the 
> day by preventing the phones from ever reaching the overflow condition (if 
> that is indeed what causes the reboots). A nightly reboot would be a 
> hackish way of fixing the problem, so I'd also love to hear if anybody has 
> any other speculations on what might be causing the spontaneous reboots. 
> We have phones in multiple buildings, and we've discovered that the 
> spontaneous reboots happen on phones on different networks, so it's not 
> specific to a particular LAN segment. For the most part, we are using the 
> standard config files for the Polycom phones and for the Linksys phones.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From wchao at yahoo.com  Thu Mar 25 09:58:15 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Thu, 25 Mar 2010 12:58:15 -0400 (EDT)
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
 phones
In-Reply-To: <71E19730-C831-4398-B042-BF8BFC4BBB04@freeswitch.org>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
	<alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>
	<71E19730-C831-4398-B042-BF8BFC4BBB04@freeswitch.org>
Message-ID: <alpine.LRH.2.00.1003251253240.5523@osgood.microoffice.com>

Brian,

No, our config files are static files on the filesystem served by Apache 
httpd (for Linksys configs and snom configs). We'll make them dynamic 
later, but first I want to solve this rebooting issue, which is pretty 
frustrating for the end users. There must be some sort of bug in Linksys 
and Polycom phones because they shouldn't spontaneously reboot. Even if 
it's a bad configuration setting or Freeswitch sending bad data, if the 
phone's firmware were solid, it ought to be able to handle those things 
gracefully rather than spontaneously rebooting. We've not had rebooting 
problems with the snom phones, which is nice.

For Polycom phones (since I'm also having rebooting problems with the 
Polycoms), we also have static configs, but served via FTP rather than 
HTTP.

Regards,
Wellie

Date: Thu, 25 Mar 2010 11:28:57 -0500
From: Brian West <brian at freeswitch.org>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
    phones

Do you generate your configs on request?
/b

On Mar 25, 2010, at 11:25 AM, Wellie Chao wrote:

      Gabe,

      My Linksys phones do seem to reboot every time they sync the config.?
      However, I have the Message_Waiting option turned off. Is there another?
      option that you think might be related to the rebooting? I do not think it?
      is a DHCP issue because the phones get the same IP address each time they?
      request one from the DHCP server. I am using the 5.1.8 firmware, which I?
      believe is the latest for the 921 and 941 phones.

      I have put one of my configuration files online here:
      http://209.190.22.48/fs/linksys/

      I changed a few things like username/password/proxy/provisioning URL, but?
      otherwise left everything the same as one of my real configuration files.

      Do you have a sample config you could send to me for a Linksys 901, 921,?
      or 941? I would love to be able to compare to see what other options might?
      make a difference.

      Regards,
      Wellie



From infos at madovsky.org  Thu Mar 25 10:01:46 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 13:01:46 -0400
Subject: [Freeswitch-users] variables scope
Message-ID: <4903B966317C4DDD9044CE93556AC17C@MOBILEE1705>

Hello, what is the scope of variables set in the gateways config ?
also is there any wiki that explains the scope of different vars ?

Thanks

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From spencer at 5ninesolutions.com  Thu Mar 25 10:16:25 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 25 Mar 2010 10:16:25 -0700
Subject: [Freeswitch-users] Fwd: Sofia crashing when registering
In-Reply-To: <191c3a031003250627r71d5a731p447b207f51fe96eb@mail.gmail.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<8FE04A55-7964-4A3B-BC94-526EF7CEDE73@5ninesolutions.com>
	<B3ACA35A-0E02-4110-88FF-D0897ACDA668@freeswitch.org>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
	<03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
	<b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>
	<191c3a031003250626h74092581g14aa4185e69d02ca@mail.gmail.com>
	<191c3a031003250627r71d5a731p447b207f51fe96eb@mail.gmail.com>
Message-ID: <64CC5F22-1D5A-4441-8AFD-24D0228F0CB6@5ninesolutions.com>

Sure, I don't mind doing the legwork.  Logging now.  Again I apologize  
for the issues before.

Spencer

On Mar 25, 2010, at 6:27 AM, Anthony Minessale wrote:

> since I forgot to mention,
>
> restart FS or at leasr mod_logfile after making the changes to the  
> config
>
>
> On Thu, Mar 25, 2010 at 8:26 AM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> If you recall, I tried to help you with this already once.
> I spent a long time on your box building out fresh trunk and  
> configuring it to trace the problem.
> Then you told me you erased it all by accident =/
>
> I suppose we can try again but I would appreciate if if you gather  
> the info for me.
>
> edit conf/autoload_configs/logfile.conf.xml
>
> look for this line:
>
> <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
> replace it with:
>
> <map name="all"  
> value="debug,info,notice,warning,err,crit,alert,console"/>
>
>
> in the cli:
>
> sofia profile internal siptrace on
> sofia loglevel all 9
>
> (repeat for any other profiles)
>
> also run wireshark or tcpdump to capture all traffic to that  
> particular provider
>
> When it happens, stop FS and the capture and get me access to the  
> resulting files.
>
> This is a rare issue triggered by some strange sip message.
> What we are trying to do here is get a copy of the traffic that  
> triggers it so we can stop it.
>
>
>
>
> On Thu, Mar 25, 2010 at 6:13 AM, Rupa Schomaker <rupa at rupa.com> wrote:
> Next time it happens, prior to a restart of the freeswitch process,  
> run the fscore_pb script to get us a set of backtraces of the  
> running process.
>
> support-d/fscore_pb gcore <yourname>
>
> try running it before hand to ensure you get expected results.   
> Getting a core does pause the process for a short period of time, so  
> do your test when someone won't notice a RTP hickup.
>
> On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <spencer at 5ninesolutions.com 
> > wrote:
> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes  
> (1-n) -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is  
> doing is bridging calls from Asterisk to the ITSP.  All media is  
> routed point to point.  The dialplan is very simple as you can  
> imagine.  If I disable registrations to Bandtel everything seems to  
> be ok.  but to my eye I don't see anything weird about the format of  
> the register that might cause a problem.
>
> Spencer
>
> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>
>> Odd nobody else has reported this behavior.  Have you viewed the  
>> wiki pages on this topic of running under a vm?
>> /b
>>
>> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>>
>>> I dunno, just thought I'd try something different.. :-)
>>>
>>> root at sip:/usr/local/freeswitch/bin# uname -r
>>> 2.6.31-20-generic
>>>
>>> I'd much rather use CentOS.  The Ubuntu machine is running under  
>>> Xen 3.4.1 if that matters.
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Thu Mar 25 10:46:36 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 13:46:36 -0400
Subject: [Freeswitch-users] variables in gateway profile
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
Message-ID: <49A0107645D242758C7C5812A49217D0@MOBILEE1705>

Are these vars can be used in a dialplan ?
how ? $${foo} ? ${foo} ? ${gateway_name[foo]} ?

Thanks
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 19, 2010 1:58 PM
  Subject: Re: [Freeswitch-users] variables in gateway profile


  yes and you can specify which direction you want to set them on, "inbound", "outbound" or "both"

  <variables>
    <variable name="foo" data="bar" direction="inbound"/>
  </variables>



  On Thu, Mar 18, 2010 at 11:55 PM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to create <variable> in gateway profiles like users profiles ?

    Thanks

    Franck

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  iax:guest at conference.freeswitch.org/888
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Thu Mar 25 10:55:43 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 12:55:43 -0500
Subject: [Freeswitch-users] variables in gateway profile
In-Reply-To: <49A0107645D242758C7C5812A49217D0@MOBILEE1705>
References: <969F9010D1ED4E48A67E3DADE50DB30D@MOBILEE1705>
	<191c3a031003191058w44834742u5c2f9e2922a5562d@mail.gmail.com>
	<49A0107645D242758C7C5812A49217D0@MOBILEE1705>
Message-ID: <D748B25E-1CB7-4CC9-A68A-BB597CAF8C7A@freeswitch.org>

$${var} is preprocess, ${var} is the one that you can use... no ${var[foo]} exists but you can do ${var:3} or ${var:0:2}

/b

On Mar 25, 2010, at 12:46 PM, Madovsky wrote:

> Are these vars can be used in a dialplan ?
> how ? $${foo} ? ${foo} ? ${gateway_name[foo]} ?
>  
> Thanks
> 

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From anthony.minessale at gmail.com  Thu Mar 25 11:00:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 13:00:37 -0500
Subject: [Freeswitch-users] variables scope
In-Reply-To: <4903B966317C4DDD9044CE93556AC17C@MOBILEE1705>
References: <4903B966317C4DDD9044CE93556AC17C@MOBILEE1705>
Message-ID: <191c3a031003251100n747f245cgec06c7908c8bac6e@mail.gmail.com>

they are the same as ordinary channel vars, they are set on the channel when
it crosses the gateway and lasts until something else overwrites it or it
hangs up

On Thu, Mar 25, 2010 at 12:01 PM, Madovsky <infos at madovsky.org> wrote:

>  Hello, what is the scope of variables set in the gateways config ?
> also is there any wiki that explains the scope of different vars ?
>
> Thanks
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From djbinter at yahoo.com  Thu Mar 25 11:06:35 2010
From: djbinter at yahoo.com (DJB)
Date: Thu, 25 Mar 2010 11:06:35 -0700 (PDT)
Subject: [Freeswitch-users] How to force a Extension use g729 only
In-Reply-To: <8a19bf2e1003250714j4c105174sb9386ed4609fa7dd@mail.gmail.com>
References: <8a19bf2e1003250714j4c105174sb9386ed4609fa7dd@mail.gmail.com>
Message-ID: <583749.89196.qm@web37508.mail.mud.yahoo.com>

http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML  - Example 5

~djbinter



________________________________
From: Aloysius Lloyd <lloyd.aloysius at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, March 25, 2010 7:14:49 AM
Subject: [Freeswitch-users] How to force a Extension use g729 only

Hi All,

I have the following setup

[IP Phones] ----[Router] --------------[FreeSWITCH] ------ [ITSP]

How to force a extension use g729 only. I could not find any parameter to set for each extension.

In Asterisk world I can do

disallow=all
allow=g729


Thanks,
Lloyd 


      
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From infos at madovsky.org  Thu Mar 25 11:11:49 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 14:11:49 -0400
Subject: [Freeswitch-users] variables scope
References: <4903B966317C4DDD9044CE93556AC17C@MOBILEE1705>
	<191c3a031003251100n747f245cgec06c7908c8bac6e@mail.gmail.com>
Message-ID: <21A705CB66D8479FBD6805EB61732612@MOBILEE1705>

ok thanks Anthony and Brian
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 2:00 PM
  Subject: Re: [Freeswitch-users] variables scope


  they are the same as ordinary channel vars, they are set on the channel when it crosses the gateway and lasts until something else overwrites it or it hangs up 


  On Thu, Mar 25, 2010 at 12:01 PM, Madovsky <infos at madovsky.org> wrote:

    Hello, what is the scope of variables set in the gateways config ?
    also is there any wiki that explains the scope of different vars ?

    Thanks



    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Mar 25 11:20:30 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 14:20:30 -0400
Subject: [Freeswitch-users] send fake UDP packets
Message-ID: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>

I use an audio stream relayed by a client that has no audio DSP card.
so when I stop to talk, no UDP audio packet is sent to FS. Do you know
a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)

Thanks

F
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From brian at freeswitch.org  Thu Mar 25 11:24:34 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 13:24:34 -0500
Subject: [Freeswitch-users] send fake UDP packets
In-Reply-To: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
Message-ID: <DEA52399-7ECE-44C8-BE42-051A6B9728CA@freeswitch.org>

This statement makes NO sense.  RTP-Timeout isn't used to hangup the call the Bye message is...  Can you elaborate on this a bit more?

/b

On Mar 25, 2010, at 1:20 PM, Madovsky wrote:

> (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)

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From anthony.minessale at gmail.com  Thu Mar 25 11:29:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 13:29:18 -0500
Subject: [Freeswitch-users] send fake UDP packets
In-Reply-To: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
Message-ID: <191c3a031003251129m1821506bqbd1a3953733b3968@mail.gmail.com>

hanging up a leg will automatically hang up b leg, if it doesnt you still
have unresolved nat problems.

make your phone register every 20 sec to keep the sip path alive.


On Thu, Mar 25, 2010 at 1:20 PM, Madovsky <infos at madovsky.org> wrote:

>  I use an audio stream relayed by a client that has no audio DSP card.
> so when I stop to talk, no UDP audio packet is sent to FS. Do you know
> a trick to send fake UDP packet to keep the call alive ? (I need to keep
> rtp-timeout short to hang up leg B when leg A hangs up)
>
> Thanks
>
> F
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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pstn:+19193869900
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From infos at madovsky.org  Thu Mar 25 11:32:13 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 14:32:13 -0400
Subject: [Freeswitch-users] send fake UDP packets
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
	<DEA52399-7ECE-44C8-BE42-051A6B9728CA@freeswitch.org>
Message-ID: <D78F26BD996141D692D364030DB43BF7@MOBILEE1705>

For now it's the only way I found to make leg B hanged up.
Apparently the Bye message is not received by the leg B when leg B when it is registered to another FS in a shared DB
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 2:24 PM
  Subject: Re: [Freeswitch-users] send fake UDP packets


  This statement makes NO sense.  RTP-Timeout isn't used to hangup the call the Bye message is...  Can you elaborate on this a bit more?


  /b


  On Mar 25, 2010, at 1:20 PM, Madovsky wrote:


    (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)




------------------------------------------------------------------------------


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Thu Mar 25 11:37:15 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 14:37:15 -0400
Subject: [Freeswitch-users] send fake UDP packets
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
	<DEA52399-7ECE-44C8-BE42-051A6B9728CA@freeswitch.org>
Message-ID: <90C89A9742D64E9DAF82A72AC0EBCC58@MOBILEE1705>

FS where leg A is registered receives well Bye message...

------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:48.472225:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:49.473166:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:51.474108:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:55.475066:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:59.476206:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:03.477236:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:07.478224:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:11.479164:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------
recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:15.480076:
------------------------------------------------------------------------
BYE sip:99999999999 at 10.10.10.138:5080;transport=udp SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g..
From: "Carl Zenmaier" <sip:8511234567890 at mydomain.com>;tag=50140543427
To: <sip:99999999999 at mydomain.com>;tag=Nt8UN6erUFK3p
Contact: <sip:8511234567890 at 10.10.10.138:64098>
CSeq: 3 BYE
Max-Forwards: 70
Call-ID: 1044306705 at 10.10.10.138

------------------------------------------------------------------------



but leg B doesn't receive anything.... I don't know yet if it's a client or FS problem. try to figure it out

  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 2:24 PM
  Subject: Re: [Freeswitch-users] send fake UDP packets


  This statement makes NO sense.  RTP-Timeout isn't used to hangup the call the Bye message is...  Can you elaborate on this a bit more?


  /b


  On Mar 25, 2010, at 1:20 PM, Madovsky wrote:


    (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)




------------------------------------------------------------------------------


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu Mar 25 11:41:37 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 14:41:37 -0400
Subject: [Freeswitch-users] send fake UDP packets
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
	<191c3a031003251129m1821506bqbd1a3953733b3968@mail.gmail.com>
Message-ID: <163E3AADF22F4688BD6D69B024748F54@MOBILEE1705>

ok.
 no nat is used. but for sure it could be a client problem
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 2:29 PM
  Subject: Re: [Freeswitch-users] send fake UDP packets


  hanging up a leg will automatically hang up b leg, if it doesnt you still have unresolved nat problems.

  make your phone register every 20 sec to keep the sip path alive.



  On Thu, Mar 25, 2010 at 1:20 PM, Madovsky <infos at madovsky.org> wrote:

    I use an audio stream relayed by a client that has no audio DSP card.
    so when I stop to talk, no UDP audio packet is sent to FS. Do you know
    a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)

    Thanks

    F

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  Twitter: http://twitter.com/FreeSWITCH_wire

  AIM: anthm
  MSN:anthony_minessale at hotmail.com
  GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
  IRC: irc.freenode.net #freeswitch

  FreeSWITCH Developer Conference
  sip:888 at conference.freeswitch.org
  googletalk:conf+888 at conference.freeswitch.org
  pstn:+19193869900



------------------------------------------------------------------------------


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From infos at madovsky.org  Thu Mar 25 12:01:36 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 15:01:36 -0400
Subject: [Freeswitch-users] international prefix
Message-ID: <F42CAF405B2C4218984A3B1F301FB1C8@MOBILEE1705>

I try to manage trunks that allow different international prefix.
for example, trunk A allows "011" and trunk B "00", and for domestic calls trunk B allows no prefix.
it's a problem when I use mod distributor since I can't gues which trunk will be used for the bridge.
I'm not sure that lcr is a good solution for this problem as it makes more DB overload.
Any idea ?

Thanks
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From anthony.minessale at gmail.com  Thu Mar 25 12:08:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 14:08:45 -0500
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <1269532290668-4798438.post@n2.nabble.com>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
Message-ID: <191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>

try latest trunk.

I added some code to wait for the video and only allow recording when video
is established.

this mod_fsv is only a test app and really we would like to get someone to
write a mp4 or some other multimedia format module.


On Thu, Mar 25, 2010 at 10:51 AM, OwenW <williams.owen at gmail.com> wrote:

>
> I get the same message when trying to playback.
>
> I've added a few debug lines - it looks as if the 'version' is never
> written
> to the file during video record.
>
> I believe this is because the following line (in mod_fsv) returns false:
> if (switch_channel_test_flag(channel, CF_VIDEO)) {
>
> Maybe I should infer from this that the channel I'm trying to record isn't
> sending video.  I'm using XLite on windows - I start a 'video call' and
> whilst the call is started I suspect it hasn't been able to negotiate video
> and perhaps fallen back to audio.  XLite does H.263 and H.263+.
>
> Owen
> --
> View this message in context:
> http://n2.nabble.com/Freeswitch-Video-Capture-and-Playback-tp4035796p4798438.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From rupa at rupa.com  Thu Mar 25 12:25:01 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 25 Mar 2010 14:25:01 -0500
Subject: [Freeswitch-users] international prefix
In-Reply-To: <F42CAF405B2C4218984A3B1F301FB1C8@MOBILEE1705>
References: <F42CAF405B2C4218984A3B1F301FB1C8@MOBILEE1705>
Message-ID: <b39273531003251225k464173ddma41d9ed00ef0a14a@mail.gmail.com>

mod_lcr is how I'd recommend dealing with this problem. <shrug/>

On Thu, Mar 25, 2010 at 2:01 PM, Madovsky <infos at madovsky.org> wrote:

>  I try to manage trunks that allow different international prefix.
> for example, trunk A allows "011" and trunk B "00", and for domestic calls
> trunk B allows no prefix.
> it's a problem when I use mod distributor since I can't gues which trunk
> will be used for the bridge.
> I'm not sure that lcr is a good solution for this problem as it makes more
> DB overload.
> Any idea ?
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From infos at madovsky.org  Thu Mar 25 12:43:03 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 15:43:03 -0400
Subject: [Freeswitch-users] international prefix
References: <F42CAF405B2C4218984A3B1F301FB1C8@MOBILEE1705>
	<b39273531003251225k464173ddma41d9ed00ef0a14a@mail.gmail.com>
Message-ID: <01A126EC18B940E89207CF789367199F@MOBILEE1705>

but it uses DB request, no ? I'd like to avoid DB request.

Thanks
  ----- Original Message ----- 
  From: Rupa Schomaker 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 3:25 PM
  Subject: Re: [Freeswitch-users] international prefix


  mod_lcr is how I'd recommend dealing with this problem. <shrug/>


  On Thu, Mar 25, 2010 at 2:01 PM, Madovsky <infos at madovsky.org> wrote:

    I try to manage trunks that allow different international prefix.
    for example, trunk A allows "011" and trunk B "00", and for domestic calls trunk B allows no prefix.
    it's a problem when I use mod distributor since I can't gues which trunk will be used for the bridge.
    I'm not sure that lcr is a good solution for this problem as it makes more DB overload.
    Any idea ?

    Thanks

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org





  -- 
  -Rupa



------------------------------------------------------------------------------


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From chris at fowler.cc  Thu Mar 25 12:48:04 2010
From: chris at fowler.cc (Chris Fowler)
Date: Thu, 25 Mar 2010 15:48:04 -0400
Subject: [Freeswitch-users] Guidance on debugging audio glitches
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C7225BE85@VMBX113.ihostexchange.net>

Hi,

Here's the problem I'm "hearing":

Test Case #1
x1001 calls x1002
- Audio is rock solid

Test Case #2
x1001 calls x3001 (Conference Bridge)
x1002 calls x3001 (Conference Bridge)
- Audio is mostly perfect.  However, occasional "glitches" in the audio can be heard

Notes:
- Bypass media is switched off.  FreeSWITCH routes audio for both the bridged, and the conferenced call
- rtp-autoflush-during-bridge is "true" on Sofia profile internal.xml
- I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see occasional auto flush of rtp packets for *both* test cases.  The glitches don't coincide with the flushing.  In fact with flushing you can't hear when data gets flushed (2x20ms packets).

Environment
- CentOS 5.4
- Tested on AWS EC2 Small, Medium, Large instances.  On Medium and Large CPU affinity for FreeSWITCH set to the 2nd core.

Questions:
- What's different between bridging a call and a conference call run through mod_conference in terms of audio processing?
- Thoughts on how to debug this?

Thanks, Chris.
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From chris at fowler.cc  Thu Mar 25 12:50:16 2010
From: chris at fowler.cc (Chris Fowler)
Date: Thu, 25 Mar 2010 15:50:16 -0400
Subject: [Freeswitch-users] Guidance on debugging audio glitches
Message-ID: <7454A296C7EDE34EA57199FAA401E2F11C7225BE88@VMBX113.ihostexchange.net>

Forgot to mention : Running trunk - build 17104

From: Chris Fowler
Sent: Thursday, March 25, 2010 12:48 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Guidance on debugging audio glitches

Hi,

Here's the problem I'm "hearing":

Test Case #1
x1001 calls x1002
- Audio is rock solid

Test Case #2
x1001 calls x3001 (Conference Bridge)
x1002 calls x3001 (Conference Bridge)
- Audio is mostly perfect.  However, occasional "glitches" in the audio can be heard

Notes:
- Bypass media is switched off.  FreeSWITCH routes audio for both the bridged, and the conferenced call
- rtp-autoflush-during-bridge is "true" on Sofia profile internal.xml
- I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see occasional auto flush of rtp packets for *both* test cases.  The glitches don't coincide with the flushing.  In fact with flushing you can't hear when data gets flushed (2x20ms packets).

Environment
- CentOS 5.4
- Tested on AWS EC2 Small, Medium, Large instances.  On Medium and Large CPU affinity for FreeSWITCH set to the 2nd core.

Questions:
- What's different between bridging a call and a conference call run through mod_conference in terms of audio processing?
- Thoughts on how to debug this?

Thanks, Chris.
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From anthony.minessale at gmail.com  Thu Mar 25 13:17:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Mar 2010 15:17:34 -0500
Subject: [Freeswitch-users] Guidance on debugging audio glitches
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C7225BE88@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C7225BE88@VMBX113.ihostexchange.net>
Message-ID: <191c3a031003251317g43ca738of00d8e0236590322@mail.gmail.com>

Did you do any debugging info on the other side?
What client is it?

I can speak from experience that I use the conference with like 10 people in
it at a minimum of 8 hours a day
and there are never any problems.

Try calling your endpoints into the FS public conference to compare

sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>

this is just a lowly VM


On Thu, Mar 25, 2010 at 2:50 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Forgot to mention : Running trunk ? build 17104
>
>
>
> *From:* Chris Fowler
> *Sent:* Thursday, March 25, 2010 12:48 PM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* Guidance on debugging audio glitches
>
>
>
> Hi,
>
>
>
> Here?s the problem I?m ?hearing?:
>
>
>
> Test Case #1
>
> x1001 calls x1002
>
> - Audio is rock solid
>
>
>
> Test Case #2
>
> x1001 calls x3001 (Conference Bridge)
>
> x1002 calls x3001 (Conference Bridge)
>
> - Audio is mostly perfect.  However, occasional ?glitches? in the audio can
> be heard
>
>
>
> Notes:
>
> - Bypass media is switched off.  FreeSWITCH routes audio for both the
> bridged, and the conferenced call
>
> - rtp-autoflush-during-bridge is ?true? on Sofia profile internal.xml
>
> - I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and
> see occasional auto flush of rtp packets for **both** test cases.  The
> glitches don?t coincide with the flushing.  In fact with flushing you can?t
> hear when data gets flushed (2x20ms packets).
>
>
>
> Environment
>
> - CentOS 5.4
>
> - Tested on AWS EC2 Small, Medium, Large instances.  On Medium and Large
> CPU affinity for FreeSWITCH set to the 2nd core.
>
>
>
> Questions:
>
> - What?s different between bridging a call and a conference call run
> through mod_conference in terms of audio processing?
>
> - Thoughts on how to debug this?
>
>
>
> Thanks, Chris.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From infos at madovsky.org  Thu Mar 25 13:26:33 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 16:26:33 -0400
Subject: [Freeswitch-users] send fake UDP packets
References: <17E8C47362104A90B439D0A9C20FD815@MOBILEE1705>
	<DEA52399-7ECE-44C8-BE42-051A6B9728CA@freeswitch.org>
Message-ID: <82603B8E9EAF4737BCA0F4413FA06F98@MOBILEE1705>

So apparently the leg B doens't receive any Bye message...
but FS receives Bye message well from legA. maybe I missed a config..

Thanks
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, March 25, 2010 2:24 PM
  Subject: Re: [Freeswitch-users] send fake UDP packets


  This statement makes NO sense.  RTP-Timeout isn't used to hangup the call the Bye message is...  Can you elaborate on this a bit more?


  /b


  On Mar 25, 2010, at 1:20 PM, Madovsky wrote:


    (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)




------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Mar 25 09:33:31 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 09:33:31 -0700
Subject: [Freeswitch-users] [Freeswitch-dev] IRC Day!
Message-ID: <87f2f3b91003250933g43da6e5rd275551e71c46f49@mail.gmail.com>

Hello all FreeSWITCHers!

We invite everyone around the world to join us in #freeswitch on
irc.freenode.net. We want everyone to be online today, even if you just
leave your IRC client running in the background. Also, if you don't have an
IRC client handy then go to http://www.freeswitch.org/ and use the web login
located on the right-hand side bar (below the features list). You'll need a
Java-enable browser.

Everyone have a great day!
-Michael
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_______________________________________________
FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org

From stevendt at primrosebank.net  Thu Mar 25 14:41:21 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Thu, 25 Mar 2010 21:41:21 -0000
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
Message-ID: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>

Hi,

is anyone using Thomson ST2030 phones with FreeSwitch ?

I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite a bit of noise (hiss) on the line and some breakup of the voice. The phones are all on a private LAN behind a router - all external calls are over the PSTN - no external IP calls are made (I don't use a VOIP provider).

I have loaded the latest Thomson SIP firmware (2.69). There are a couple of options on the phone that I've tried such as Silence Suppression, Acoustic Echo Cancellation and Packet Loss Compensation with no improvement. Most of the other options are beyond me I'm afraid, so I'd be really interested to hear from anyone who has good (or even acceptable !) results from the ST2030 - particularly if they could share their phone configuration

regards
Dave
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From matt at packetfoundry.net  Thu Mar 25 14:47:17 2010
From: matt at packetfoundry.net (Matt Putnam)
Date: Thu, 25 Mar 2010 16:47:17 -0500 (CDT)
Subject: [Freeswitch-users] Per User Music on Hold
Message-ID: <1dbb01cacc65$723d8cc0$56b8a640$@net>

Is there a way to change music on hold per user instead of per channel? I
have one user that I would like to have different hold music than the
default. I can get it to work outbound by setting it on the channel but
inbound I cant seem to find a way to change its always the default
specified in vars.xml. 

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From vetali100 at gmail.com  Thu Mar 25 15:13:39 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 26 Mar 2010 00:13:39 +0200
Subject: [Freeswitch-users] Media traffic to bypass FreeSWITCH and to be
	exchanged by peers only
Message-ID: <a858e7691003251513l298b27b7jb2af934dc8fe815a@mail.gmail.com>

Hi,

Very newbie question:

Currently both SIP and RTP traffic passes thru the FreeSWITCH server.

            Client A -->SIP and RTP -->FS server-->Client B

Is it possible that only SIP traffic will pass thru FS, and media will be
exchanged between Client A and Client B directly?

            Client A -->SIP -->FS server-->Client B
            Client A -->RTP -->Client B

How can I configure this?

Sure both client A and B will use same codec, otherwise FS Server is a must
for transcoding, right? Or is a must for anything else...

Thank you,
vIT
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From brian at freeswitch.org  Thu Mar 25 15:18:20 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 17:18:20 -0500
Subject: [Freeswitch-users] Media traffic to bypass FreeSWITCH and to be
	exchanged by peers only
In-Reply-To: <a858e7691003251513l298b27b7jb2af934dc8fe815a@mail.gmail.com>
References: <a858e7691003251513l298b27b7jb2af934dc8fe815a@mail.gmail.com>
Message-ID: <B7EF57F7-39AF-41D4-B873-D42B02B2D10C@freeswitch.org>

http://wiki.freeswitch.org/wiki/Bypass_media

/b

On Mar 25, 2010, at 5:13 PM, Vitalii Colosov wrote:

> Hi,
> 
> Very newbie question:
> 
> Currently both SIP and RTP traffic passes thru the FreeSWITCH server.
> 
>             Client A -->SIP and RTP -->FS server-->Client B
> 
> Is it possible that only SIP traffic will pass thru FS, and media will be exchanged between Client A and Client B directly?
> 
>             Client A -->SIP -->FS server-->Client B
>             Client A -->RTP -->Client B
> 
> How can I configure this?
> 
> Sure both client A and B will use same codec, otherwise FS Server is a must for transcoding, right? Or is a must for anything else...
> 
> Thank you,
> vIT
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Thu Mar 25 15:27:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 25 Mar 2010 18:27:11 -0400
Subject: [Freeswitch-users] international prefix
In-Reply-To: <01A126EC18B940E89207CF789367199F@MOBILEE1705>
References: <F42CAF405B2C4218984A3B1F301FB1C8@MOBILEE1705>
	<b39273531003251225k464173ddma41d9ed00ef0a14a@mail.gmail.com>
	<01A126EC18B940E89207CF789367199F@MOBILEE1705>
Message-ID: <945A4F2A-404B-4883-9C30-B55065B3720D@jerris.com>

dialplan can do all of this trivially

On Mar 25, 2010, at 3:43 PM, Madovsky wrote:

> but it uses DB request, no ? I'd like to avoid DB request.
>  
> Thanks
> ----- Original Message -----
> From: Rupa Schomaker
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, March 25, 2010 3:25 PM
> Subject: Re: [Freeswitch-users] international prefix
> 
> mod_lcr is how I'd recommend dealing with this problem. <shrug/>
> 
> On Thu, Mar 25, 2010 at 2:01 PM, Madovsky <infos at madovsky.org> wrote:
> I try to manage trunks that allow different international prefix.
> for example, trunk A allows "011" and trunk B "00", and for domestic calls trunk B allows no prefix.
> it's a problem when I use mod distributor since I can't gues which trunk will be used for the bridge.
> I'm not sure that lcr is a good solution for this problem as it makes more DB overload.
> Any idea ?
>  
> Thanks

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From vetali100 at gmail.com  Thu Mar 25 15:30:17 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Fri, 26 Mar 2010 00:30:17 +0200
Subject: [Freeswitch-users] Media traffic to bypass FreeSWITCH and to be
	exchanged by peers only
In-Reply-To: <B7EF57F7-39AF-41D4-B873-D42B02B2D10C@freeswitch.org>
References: <a858e7691003251513l298b27b7jb2af934dc8fe815a@mail.gmail.com>
	<B7EF57F7-39AF-41D4-B873-D42B02B2D10C@freeswitch.org>
Message-ID: <a858e7691003251530r44e1d306ue03d0d93ecf407d9@mail.gmail.com>

Thanks Brian,

Now I remember - I 've read this article long time back, but at that time I
did not have any clue what it is about, so it was quickly forgotten :-)
Thanks for pointing this out.

BTW, i tried to use it and did not hear voice, but probably this is already
NAT issue, as mentioned also in the article...

vIT


2010/3/26 Brian West <brian at freeswitch.org>

> http://wiki.freeswitch.org/wiki/Bypass_media
>
> /b
>
> On Mar 25, 2010, at 5:13 PM, Vitalii Colosov wrote:
>
> > Hi,
> >
> > Very newbie question:
> >
> > Currently both SIP and RTP traffic passes thru the FreeSWITCH server.
> >
> >             Client A -->SIP and RTP -->FS server-->Client B
> >
> > Is it possible that only SIP traffic will pass thru FS, and media will be
> exchanged between Client A and Client B directly?
> >
> >             Client A -->SIP -->FS server-->Client B
> >             Client A -->RTP -->Client B
> >
> > How can I configure this?
> >
> > Sure both client A and B will use same codec, otherwise FS Server is a
> must for transcoding, right? Or is a must for anything else...
> >
> > Thank you,
> > vIT
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Thu Mar 25 15:33:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 17:33:16 -0500
Subject: [Freeswitch-users] Media traffic to bypass FreeSWITCH and to be
	exchanged by peers only
In-Reply-To: <a858e7691003251530r44e1d306ue03d0d93ecf407d9@mail.gmail.com>
References: <a858e7691003251513l298b27b7jb2af934dc8fe815a@mail.gmail.com>
	<B7EF57F7-39AF-41D4-B873-D42B02B2D10C@freeswitch.org>
	<a858e7691003251530r44e1d306ue03d0d93ecf407d9@mail.gmail.com>
Message-ID: <73BAAE7C-8181-4FC6-AE2A-0E64C64074AD@freeswitch.org>

It won't work very well with nat.

/b

On Mar 25, 2010, at 5:30 PM, Vitalii Colosov wrote:

> Thanks Brian,
> 
> Now I remember - I 've read this article long time back, but at that time I did not have any clue what it is about, so it was quickly forgotten :-)
> Thanks for pointing this out.
> 
> BTW, i tried to use it and did not hear voice, but probably this is already NAT issue, as mentioned also in the article...
> 
> vIT
> 



From jcasale at activenetwerx.com  Thu Mar 25 15:55:34 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 25 Mar 2010 22:55:34 +0000
Subject: [Freeswitch-users] Disabling mod_xml_rpc
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>

If the web interface is not needed, can this module be left unloaded
or do other services depend on it?

Thanks,
jlc


From brian at freeswitch.org  Thu Mar 25 16:02:36 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 18:02:36 -0500
Subject: [Freeswitch-users] Disabling mod_xml_rpc
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>
Message-ID: <E953579C-B73E-463A-841F-C8F4173BE731@freeswitch.org>

Thats the point of modules.  You can choose to not load it.

/b

On Mar 25, 2010, at 5:55 PM, Joseph L. Casale wrote:

> If the web interface is not needed, can this module be left unloaded
> or do other services depend on it?
> 
> Thanks,
> jlc



From freeswitch.org at todandlorna.com  Thu Mar 25 16:10:19 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Thu, 25 Mar 2010 17:10:19 -0600
Subject: [Freeswitch-users] Disabling mod_xml_rpc
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>
Message-ID: <4BABED5B.2000809@todandlorna.com>

Just comment it out in the modules.conf.xml file.  It will be fine.

-Tod Hansmann

On 3/25/2010 4:55 PM, Joseph L. Casale wrote:
> If the web interface is not needed, can this module be left unloaded
> or do other services depend on it?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    


From msc at freeswitch.org  Thu Mar 25 16:10:56 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 25 Mar 2010 16:10:56 -0700
Subject: [Freeswitch-users] Per User Music on Hold
In-Reply-To: <1dbb01cacc65$723d8cc0$56b8a640$@net>
References: <1dbb01cacc65$723d8cc0$56b8a640$@net>
Message-ID: <87f2f3b91003251610k53fbdf8ck9b7bf38e4e4f8fa9@mail.gmail.com>

On inbound calls you'll probably need to do a user_data lookup:
<action application="set" data="user_moh=${user_data $dialed_number@${domain}
var moh_var"/>

Be sure to change "moh_var" to whatever variable name you use. You should
also do an eval in there somewhere so that if the ${user_moh} variable is
empty then it can just use the default.

-MC

On Thu, Mar 25, 2010 at 2:47 PM, Matt Putnam <matt at packetfoundry.net> wrote:

>  Is there a way to change music on hold per user instead of per channel? I
> have one user that I would like to have different hold music than the
> default. I can get it to work outbound by setting it on the channel but
> inbound I cant seem to find a way to change its always the default specified
> in vars.xml.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jcasale at activenetwerx.com  Thu Mar 25 16:14:13 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 25 Mar 2010 23:14:13 +0000
Subject: [Freeswitch-users] Disabling mod_xml_rpc
In-Reply-To: <E953579C-B73E-463A-841F-C8F4173BE731@freeswitch.org>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EA33A@prato.activenetwerx.local>
	<E953579C-B73E-463A-841F-C8F4173BE731@freeswitch.org>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EA46F@prato.activenetwerx.local>

>Thats the point of modules.  You can choose to not load it.

I got that:) I was curious if anything else depends on it, or
just the webserver?


From rupa at rupa.com  Thu Mar 25 16:53:36 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 25 Mar 2010 18:53:36 -0500
Subject: [Freeswitch-users] Guidance on debugging audio glitches
In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11C7225BE85@VMBX113.ihostexchange.net>
References: <7454A296C7EDE34EA57199FAA401E2F11C7225BE85@VMBX113.ihostexchange.net>
Message-ID: <b39273531003251653g863b204w4bd3eb7b03334715@mail.gmail.com>

EC2 - you are running under a VM.  Perhaps you are getting glitches due to
timing issues that are more severe when the audio is being mixed as apposed
to just passing the audio between endpoints?

On Thu, Mar 25, 2010 at 2:48 PM, Chris Fowler <chris at fowler.cc> wrote:

>  Hi,
>
>
>
> Here?s the problem I?m ?hearing?:
>
>
>
> Test Case #1
>
> x1001 calls x1002
>
> - Audio is rock solid
>
>
>
> Test Case #2
>
> x1001 calls x3001 (Conference Bridge)
>
> x1002 calls x3001 (Conference Bridge)
>
> - Audio is mostly perfect.  However, occasional ?glitches? in the audio can
> be heard
>
>
>
> Notes:
>
> - Bypass media is switched off.  FreeSWITCH routes audio for both the
> bridged, and the conferenced call
>
> - rtp-autoflush-during-bridge is ?true? on Sofia profile internal.xml
>
> - I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and
> see occasional auto flush of rtp packets for **both** test cases.  The
> glitches don?t coincide with the flushing.  In fact with flushing you can?t
> hear when data gets flushed (2x20ms packets).
>
>
>
> Environment
>
> - CentOS 5.4
>
> - Tested on AWS EC2 Small, Medium, Large instances.  On Medium and Large
> CPU affinity for FreeSWITCH set to the 2nd core.
>
>
>
> Questions:
>
> - What?s different between bridging a call and a conference call run
> through mod_conference in terms of audio processing?
>
> - Thoughts on how to debug this?
>
>
>
> Thanks, Chris.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From gavin.henry at gmail.com  Thu Mar 25 16:53:56 2010
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 25 Mar 2010 23:53:56 +0000
Subject: [Freeswitch-users] call info display
In-Reply-To: <94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
	<191c3a031003160848y3a238fadsdcd4ae654dacc1d6@mail.gmail.com>
	<3B6C614E-0F72-40CB-AA54-31932347233B@ukgb.net>
	<191c3a031003161011n52e40f72w6824b95e3c66ad36@mail.gmail.com>
	<94C53276-3AC9-4AA2-8C50-3EFDBAADA0F2@ukgb.net>
Message-ID: <13ca621c1003251653w1caad2a4h3e3597b12387dbe@mail.gmail.com>

Hi Ken,

This is just dialplan stuff. Match an inbound DDI, use the callerid
variable and prefix it with the customer name. The phones just display
what the callerid is the *you* send it.

How you match all this can be done a millions ways; via an DB, LDAP,
google search whatever. If you can think of it it can be done.

Check the wiki and samples out.

Thanks,

Gavin.

On 16/03/2010, Ken Gillett <ken at ukgb.net> wrote:
> Previously I had a company with a call centre, in which we used a Meridian
> Option 11 (hardware telephone switch) on an ISDN 30 trunk with a range of
> DDI numbers. Each client was provided with one of those DDI numbers on which
> their customers would call us, so it was vital the operator knew which
> client's customer was calling. Within the configuration of the Option 11 I
> named each incoming DDI with the respective client's (Company) name and then
> that name was sent to the extension, along with the incoming CLI, so each
> operator knew which client's number had been called (and hence how to
> respond) and also who was calling, either their number, or name if it could
> be matched in an address directory (although cannot now remember if that was
> done by the switch itself or the extension phone).
>
> Nothing strange about any of that, but I'm trying to relate those same
> requirements to the capabilities of FS. My assumption has been that FS would
> be able to match anything traditional telephone switches could manage, but I
> now need some confirmation of this.
>
> So, are the callee ID and number the incoming line name and number
> respectively and it's those that FS sends to the extension for display?
>
> Where does this data come from? Surely naming the incoming lines must be
> part of the configuration.
>
> Can I name them to whatever I want?
>
> Is the incoming CLI number sent directly to the extension (where it's up to
> the extension what it then displays)?
> Or
> Does FS do any matching to a name in a directory first and then send the
> matched name to the extension?
>
> Sorry to keep asking - I'm sure this is very obvious to those who know, but
> I'm still trying to learn all about FS and what it can and can't do.
>
>
> On 16 Mar 2010, at 17:11, Anthony Minessale wrote:
>
>> as i said it is phone specific.
>>
>> The callee id and caleeid number is sent
>>
>> on polycom and astra via sip UPDATE
>> on snom and FS->FS via INFO
>>
>>
>> There is no configuration necessary, its the default behavior.
>>
>>
>> On Tue, Mar 16, 2010 at 11:23 AM, Ken Gillett <ken at ukgb.net> wrote:
>> Thanks, but could I ask for more specifics? Exactly what info is sent and
>> is it sent as part of the SIP info, i.e. so any SIP device ought to
>> display it correctly? Are there any docs I can read that will explain in
>> detail about the information that is sent (i.e. call info, not SIP
>> protocol stuff) and how to configureFS to send exactly what is required?
>>
>>
>> On 16 Mar 2010, at 15:48, Anthony Minessale wrote:
>>
>> > depending on the phone FreeSWITCH will update the display any time a
>> > call is bridged.
>> >
>> > it works on polycom,snom and aastra and I am not sure if there are more.
>> >
>> >
>> > On Tue, Mar 16, 2010 at 5:50 AM, Ken Gillett <ken at ukgb.net> wrote:
>> >> Sorry to ask what are perhaps very basic questions, but could someone
>> >> please enlighten me on FreeSwitch's capabilities with regard to
>> >> information for display on the recipient's extension.
>> >>
>> >> When a call is received, the extension really needs to display
>> >> information about the origin and the intended recipient of the call.
>> >> Normally for an incoming external call, the former is supplied by the
>> >> 'incoming CLI' (often linked to locally stored name) and the latter
>> >> probably by associating a name with each incoming line, so in this way
>> >> the recipient can see at a glance who called whom (or what).
>> >>
>> >> Am I right in thinking that within SIP there is the provision to send
>> >> information about the calling party (origin), so the called party
>> >> (recipient) doesn't have to rely on CLI and matching numbers to locally
>> >> stored directories?
>> >>
>> >> So, how does FS handle all the above?
>> >> What does it pass to the recipient, for both external and internal
>> >> calls?
>> >> Is this part of FS configuration? IOW, do you have programmatic access
>> >> to such information and can you specify exactly what is sent to an
>> >> extension?
>> >
>
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

http://www.suretecsystems.com/services/openldap/
http://www.suretectelecom.com


From nandy1925 at gmail.com  Thu Mar 25 17:44:30 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 26 Mar 2010 08:44:30 +0800
Subject: [Freeswitch-users] call info display
In-Reply-To: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
References: <7A4674AD-A840-4506-A36F-DB8E822C0E72@ukgb.net>
Message-ID: <7d0bfd8c1003251744s3527f7daw1d95543db3789e83@mail.gmail.com>

ken,

i'll share to you my experience. i hv an FXO. for incoming calls, the
CallerID number, received after the 1st ring, is sent to Freeswitch during
SIP invitation. there's a mod_cidlookup module in freeswitch if u want to
get the caller's name - and decide what to do w/ the incoming call.

-nandy


On Tue, Mar 16, 2010 at 6:50 PM, Ken Gillett <ken at ukgb.net> wrote:

> Sorry to ask what are perhaps very basic questions, but could someone
> please enlighten me on FreeSwitch's capabilities with regard to information
> for display on the recipient's extension.
>
> When a call is received, the extension really needs to display information
> about the origin and the intended recipient of the call. Normally for an
> incoming external call, the former is supplied by the 'incoming CLI' (often
> linked to locally stored name) and the latter probably by associating a name
> with each incoming line, so in this way the recipient can see at a glance
> who called whom (or what).
>
> Am I right in thinking that within SIP there is the provision to send
> information about the calling party (origin), so the called party
> (recipient) doesn't have to rely on CLI and matching numbers to locally
> stored directories?
>
> So, how does FS handle all the above?
> What does it pass to the recipient, for both external and internal calls?
> Is this part of FS configuration? IOW, do you have programmatic access to
> such information and can you specify exactly what is sent to an extension?
>
>
> Ken G i l l e t t
>
> _/_/_/_/_/_/_/_/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Thu Mar 25 20:07:21 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Mar 2010 23:07:21 -0400
Subject: [Freeswitch-users] new users in directory
Message-ID: <332BD48DAA2D4E5398B80C1F8E6470EB@MOBILEE1705>

good evening,

is it really necessary to do a "reloadxml" everytime a new user is created ?

Thanks
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From brian at freeswitch.org  Thu Mar 25 20:25:48 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 25 Mar 2010 22:25:48 -0500
Subject: [Freeswitch-users] new users in directory
In-Reply-To: <332BD48DAA2D4E5398B80C1F8E6470EB@MOBILEE1705>
References: <332BD48DAA2D4E5398B80C1F8E6470EB@MOBILEE1705>
Message-ID: <2839F103-07A2-4D73-8C0D-B687E821DCBE@freeswitch.org>

If you use mod_xml_curl it isn't.

/b

On Mar 25, 2010, at 10:07 PM, Madovsky wrote:

> good evening,
>  
> is it really necessary to do a "reloadxml" everytime a new user is created ?
>  
> Thanks
> ______

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From gkuri at ieee.org  Thu Mar 25 20:43:19 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Thu, 25 Mar 2010 20:43:19 -0700
Subject: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
	phones
In-Reply-To: <alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>
References: <alpine.LRH.2.00.1003231324360.5523@osgood.microoffice.com>
	<8b1c9cda1003242042l6d6e5bfr3a75723d7fb5cc1@mail.gmail.com>
	<alpine.LRH.2.00.1003251218420.5523@osgood.microoffice.com>
Message-ID: <8b1c9cda1003252043j4ea1333dq84ddb4277cec9c8a@mail.gmail.com>

Wellie,

Line 51 of the config I pulled off your site, has Message_Waiting ...

       <Message_Waiting_1_ ua="na">No</Message_Waiting_1_>

Remove this line and I bet your problem will go away.

I can't think of anything else off-hand that would cause the phones to
reboot due to syncing the config.

Cheers,
Gabe

On Thu, Mar 25, 2010 at 9:25 AM, Wellie Chao <wchao at yahoo.com> wrote:
> Gabe,
>
> My Linksys phones do seem to reboot every time they sync the config.
> However, I have the Message_Waiting option turned off. Is there another
> option that you think might be related to the rebooting? I do not think it
> is a DHCP issue because the phones get the same IP address each time they
> request one from the DHCP server. I am using the 5.1.8 firmware, which I
> believe is the latest for the 921 and 941 phones.
>
> I have put one of my configuration files online here:
> http://209.190.22.48/fs/linksys/
>
> I changed a few things like username/password/proxy/provisioning URL, but
> otherwise left everything the same as one of my real configuration files.
>
> Do you have a sample config you could send to me for a Linksys 901, 921,
> or 941? I would love to be able to compare to see what other options might
> make a difference.
>
> Regards,
> Wellie
>
>
> Date: Wed, 24 Mar 2010 20:42:38 -0700
> From: Gabriel Kuri <gkuri at ieee.org>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Spontaneous reboots of Polycom and Linksys
> ? ?phones
>
> If you're using central provisioning with the Linksys/Cisco Small
> Business series phones and you leave the Message_Waiting option in the
> config (which is the default, very stupid), the phones will reboot
> every time they sync the config, if they have a voicemail waiting.
>
> Check to see if the Linksys spontaneous reboots correlate with pulling
> down the config (assuming you're using central provisioning) and make
> sure you either comment or delete Message_Waiting from the config.
>
> You may also want to see if there's any correlation with DHCP and the
> phones renewing their IP. We put all of our phones on DHCP
> reservations, so their IP doesn't/shouldn't change. I'm not sure what
> the phone will do if it goes to renew its address and the DHCP server
> hands it a different IP, it may cause the phone to reboot?
>
> Cheers,
> Gabe
>
>
> On Wed, Mar 24, 2010 at 6:35 PM, Wellie Chao <wchao at yahoo.com> wrote:
>> I am using Freeswitch and I have some phones that spontaneously reboot
>> (often in the middle of a call).
>>
>> I did some searching around on the web, but didn't find any information
>> that really applied to the problem I'm experiencing. I have a few dozen
>> Polycom phones and a handful of Linksys phones of various models: Polycom
>> IP301, IP320, IP331, IP430, IP501, IP601; also Linksys SPA901, SPA921,
>> SPA941.
>>
>> I also have snom phones, but they don't reboot spontaneously. It's only
>> the Polycom and Linksys phones that reboot spontaneously, and it seems to
>> be correlated to call volume. Heavy usage of the Polycom and Linksys
>> phones causes them to reboot more frequently (i.e. shorter duration
>> between the reboots). For people making 50 calls a day, the Polycom phones
>> spontaneously reboot every three days or so (and always on a call, it
>> seems). For people making one or two calls a day, the Polycom will go
>> months without a spontaneous reboot. This correlation between call volume
>> and reboot frequency leads me to believe that there is some bug in the
>> Polycom firmware, perhaps some of overflow condition that accumulates over
>> time and eventually causes a crash (and subsequent reboot).
>>
>> I was wondering if anybody else experienced something similar, and whether
>> anyone had developed a solution. One thing I am considering is to set up a
>> script to reboot the Polycom phones in the middle of the night every day,
>> which would hopefully prevent them from spontaneously rebooting during the
>> day by preventing the phones from ever reaching the overflow condition (if
>> that is indeed what causes the reboots). A nightly reboot would be a
>> hackish way of fixing the problem, so I'd also love to hear if anybody has
>> any other speculations on what might be causing the spontaneous reboots.
>> We have phones in multiple buildings, and we've discovered that the
>> spontaneous reboots happen on phones on different networks, so it's not
>> specific to a particular LAN segment. For the most part, we are using the
>> standard config files for the Polycom phones and for the Linksys phones.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From yehavi.bourvine at gmail.com  Thu Mar 25 21:59:06 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 26 Mar 2010 06:59:06 +0200
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
In-Reply-To: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>
Message-ID: <d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com>

We have one ST2030 and one other model of them which we bought for
evaluation. We have the same problem as you describe, both with Freeswitch
and Asterisk. That's why we have only one unit and not more...

from my experience, you get what you pay for.

                  __Yehavi:

 2010/3/25 Dave Stevenson <stevendt at primrosebank.net>

>  Hi,
>
> is anyone using Thomson ST2030 phones with FreeSwitch ?
>
> I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the
> Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite
> a bit of noise (hiss) on the line and some breakup of the voice. The phones
> are all on a private LAN behind a router - all external calls are over the
> PSTN - no external IP calls are made (I don't use a VOIP provider).
>
> I have loaded the latest Thomson SIP firmware (2.69). There are a couple of
> options on the phone that I've tried such as Silence Suppression, Acoustic
> Echo Cancellation and Packet Loss Compensation with no improvement. Most of
> the other options are beyond me I'm afraid, so I'd be really interested to
> hear from anyone who has good (or even acceptable !) results from the ST2030
> - particularly if they could share their phone configuration
>
> regards
> Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gcd at i.ph  Thu Mar 25 22:49:44 2010
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 26 Mar 2010 13:49:44 +0800
Subject: [Freeswitch-users] GSM gateway recommendation
Message-ID: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>

hello everybody,

i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
brand/model you had and your go/no-go recommendation.

tks a lot,
nandy
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From itamar at ispbrasil.com.br  Fri Mar 26 02:01:44 2010
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 26 Mar 2010 06:01:44 -0300
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
Message-ID: <d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>

On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon <gcd at i.ph> wrote:
> hello everybody,
>
> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> brand/model you had and your go/no-go recommendation.
>
> tks a lot,
> nandy
>

portech ?

-- 
------------

Itamar Reis Peixoto

e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From williams.owen at gmail.com  Fri Mar 26 02:35:12 2010
From: williams.owen at gmail.com (OwenW)
Date: Fri, 26 Mar 2010 01:35:12 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
	<191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
Message-ID: <1269596112846-4802883.post@n2.nabble.com>


Thanks Anthony,

I'm very impressed with the quick response.

The additional waiting/logging makes things clearer.  Your new code is
waiting for the video and never getting it.  In my case this is correct
behaviour as X-Lite and FreeSwitch are not able to negotiate video furing
offer/answer.

Here is the SDP from XLite Invite:

v=0
o=- 6 2 IN IP4 192.168.30.10
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.30.10
t=0 0
m=audio 46602 RTP/AVP 107 0 8 101
a=sendrecv
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : oxkiVo6c sUtX2luZ 192.168.30.10 46602
m=video 37220 RTP/AVP 115 34
a=sendrecv
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1
a=alt:1 1 : OFwEAssN b3l2u5lG 192.168.30.10 37220

And here is the SDP from freeswitch in the response:
v=0
o=FreeSWITCH 1269574772 1269574773 IN IP4 192.168.30.10
s=FreeSWITCH
c=IN IP4 192.168.30.10
t=0 0
m=audio 20138 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 0 RTP/AVP 19

XLite is offering video H.263 and H.263(1998) but Freeswitch is answering
with a zero port (RTP/AVP 19).  The only thing that is being successfully
negotiated is the audio (PCMU/8000).

Can you confirm whether mod_fsv can record H.263?  Do you know how I can
pursuade freeswitch to accept the video?  I'm a little new to freeswitch - I
suspect mod_H26x may be part of the answer however there isn't any
documentation...

Owen



-- 
View this message in context: http://n2.nabble.com/Freeswitch-Video-Capture-and-Playback-tp4035796p4802883.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From williams.owen at gmail.com  Fri Mar 26 02:54:06 2010
From: williams.owen at gmail.com (OwenW)
Date: Fri, 26 Mar 2010 01:54:06 -0800 (PST)
Subject: [Freeswitch-users] How To: Sip Logging and Freeswitch
Message-ID: <1269597246475-4802944.post@n2.nabble.com>


This is a quick guide as to one approach to logging the sip messages between
a Sip phone and freeswitch.  There are other methods (e.g. wireshark) - feel
free to reply if you have other alternatives.  

It involves putting a java TCP Proxy (grinder) in front of freeswitch and
reconfiguring your SipPhone to send SIP to the proxy.  

The software (grinder) is Java based so will require an JDK.

Once you've downloaded Grinder create a file called startProxy5061.cmd and
put the following into it:
call setGrinderEnv.cmd
java -cp %CLASSPATH% net.grinder.TCPProxy -console -localhost 192.168.30.10
-localport 5061 -remotehost 192.168.30.10 -remoteport 5060 >
freeswitch_sip.log

The proxy forwards all traffic it receives on 5061 to 5060 and logs to
freeswitch_sip.log.

You'll need to change the IPs above.
Create another file called setGrinderEnv.cmd and put the following in to it:
set GRINDERPATH=C:\dev\apps\grinder-3.3
set GRINDERPROPERTIES=C:\dev\apps\grinder-3.3\grinder.properties
set CLASSPATH=%GRINDERPATH%\lib\grinder.jar;%CLASSPATH%
set JAVA_HOME=c:\Program Files\java\jdk1.6.0_11
PATH=%JAVA_HOME%\bin;%PATH%

You'll need to change the paths above.

The soft phone will need to be configured to use port 5061 instead of 5060
as the proxy.

Owen



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From williams.owen at gmail.com  Fri Mar 26 02:57:21 2010
From: williams.owen at gmail.com (OwenW)
Date: Fri, 26 Mar 2010 01:57:21 -0800 (PST)
Subject: [Freeswitch-users] How To: Sip Logging and Freeswitch
In-Reply-To: <1269597246475-4802944.post@n2.nabble.com>
References: <1269597246475-4802944.post@n2.nabble.com>
Message-ID: <1269597441480-4802952.post@n2.nabble.com>


Oops here's the link to the java proxy software.  

http://grinder.sourceforge.net/
-- 
View this message in context: http://n2.nabble.com/How-To-Sip-Logging-and-Freeswitch-tp4802944p4802952.html
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From nazim.agabekov at gmail.com  Fri Mar 26 03:03:39 2010
From: nazim.agabekov at gmail.com (Nazim Agabekov)
Date: Fri, 26 Mar 2010 14:03:39 +0400
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
Message-ID: <4BAC867B.9010503@gmail.com>

Hello Nandy

I'm using Sangoma's SS7 gateway for interconnecting E1/ISUP and
FreeSWITCH. It's rock solid and easy maintain.
Also, Sangoma's support team is  _Very_ friendly.

http://sangoma.com/products/software_building_blocks/ss7_solutions/ss7_signaling_and_media_gateway.html

On 03/26/2010 09:49 AM, Nandy Dagondon wrote:
> hello everybody,
>
> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share
> what  brand/model you had and your go/no-go recommendation.
>
> tks a lot,
> nandy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   

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From devel at thom.fr.eu.org  Fri Mar 26 03:15:12 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Fri, 26 Mar 2010 11:15:12 +0100
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <1269596112846-4802883.post@n2.nabble.com>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
	<191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
	<1269596112846-4802883.post@n2.nabble.com>
Message-ID: <3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org>

You need to add the H26x codec in freeswitch available codecs list
(global_codec_prefs or outbound_codec_prefs, check in the wiki the one that
applies here).

Fran?ois

On Fri, 26 Mar 2010 01:35:12 -0800 (PST), OwenW <williams.owen at gmail.com>
wrote:
> Thanks Anthony,
> 
> I'm very impressed with the quick response.
> 
> The additional waiting/logging makes things clearer.  Your new code is
> waiting for the video and never getting it.  In my case this is correct
> behaviour as X-Lite and FreeSwitch are not able to negotiate video
furing
> offer/answer.
> 
> Here is the SDP from XLite Invite:
> 
> v=0
> o=- 6 2 IN IP4 192.168.30.10
> s=CounterPath X-Lite 4.0
> c=IN IP4 192.168.30.10
> t=0 0
> m=audio 46602 RTP/AVP 107 0 8 101
> a=sendrecv
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 1 : oxkiVo6c sUtX2luZ 192.168.30.10 46602
> m=video 37220 RTP/AVP 115 34
> a=sendrecv
> a=rtpmap:115 H263-1998/90000
> a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
> a=rtpmap:34 H263/90000
> a=fmtp:34 QCIF=1;CIF=1
> a=alt:1 1 : OFwEAssN b3l2u5lG 192.168.30.10 37220
> 
> And here is the SDP from freeswitch in the response:
> v=0
> o=FreeSWITCH 1269574772 1269574773 IN IP4 192.168.30.10
> s=FreeSWITCH
> c=IN IP4 192.168.30.10
> t=0 0
> m=audio 20138 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> m=video 0 RTP/AVP 19
> 
> XLite is offering video H.263 and H.263(1998) but Freeswitch is
answering
> with a zero port (RTP/AVP 19).  The only thing that is being
successfully
> negotiated is the audio (PCMU/8000).
> 
> Can you confirm whether mod_fsv can record H.263?  Do you know how I can
> pursuade freeswitch to accept the video?  I'm a little new to freeswitch
-
> I
> suspect mod_H26x may be part of the answer however there isn't any
> documentation...
> 
> Owen


From yivzhenko at mksat.net  Fri Mar 26 04:23:49 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Fri, 26 Mar 2010 13:23:49 +0200
Subject: [Freeswitch-users] Bad Video in latest SVN
Message-ID: <201003261323.50280.yivzhenko@mksat.net>

Hi all,

After upgrade from SVN 16013 to latest SVN 17109
Video quality become very bad. 
I'm use H264 video codec and Grandstream GXV3140 phones 

Anyone knows why this may happen?





From williams.owen at gmail.com  Fri Mar 26 07:25:32 2010
From: williams.owen at gmail.com (OwenW)
Date: Fri, 26 Mar 2010 06:25:32 -0800 (PST)
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
	<191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
	<1269596112846-4802883.post@n2.nabble.com>
	<3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org>
Message-ID: <1269613532826-4804168.post@n2.nabble.com>


Thanks that works but video is very choppy.  I get lots of artefacts even if
I move my head even slightly.  It looks like its going through a very small
pipe.

Freeswitch is returning the following SDP:
v=0
o=FreeSWITCH 1269580573 1269580574 IN IP4 192.168.30.10
s=FreeSWITCH
c=IN IP4 192.168.30.10
t=0 0
m=audio 30316 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 17834 RTP/AVP 115
a=rtpmap:115 H263-1998/90000

I've tried doing things on the soft phone - I can increase the send bit rate
and reduce the definition but its still choppy.

Is this some way to increase the bit-rate on freeswitch side?  

Owen




I update vars.xml with:
 <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H263,H263-1998"/>

I can record and playback video using mod_fsv (thanks Anthony!).  

It's very choppy though.
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From brian at freeswitch.org  Fri Mar 26 07:32:50 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Mar 2010 09:32:50 -0500
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <1269613532826-4804168.post@n2.nabble.com>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
	<191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
	<1269596112846-4802883.post@n2.nabble.com>
	<3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org>
	<1269613532826-4804168.post@n2.nabble.com>
Message-ID: <FEFF4F1E-3493-448A-B94B-E95BC18058DC@freeswitch.org>

No because FreeSWITCH doesn't encode the video... its returning exactly what was received packet by packet.  So to clarify you sent crappy video to FreeSWITCH, It wrote it down, You called back and viewed the crappy video coming back at you.  :P

/b

On Mar 26, 2010, at 9:25 AM, OwenW wrote:

> 
> Thanks that works but video is very choppy.  I get lots of artefacts even if
> I move my head even slightly.  It looks like its going through a very small
> pipe.
> 
> Freeswitch is returning the following SDP:
> v=0
> o=FreeSWITCH 1269580573 1269580574 IN IP4 192.168.30.10
> s=FreeSWITCH
> c=IN IP4 192.168.30.10
> t=0 0
> m=audio 30316 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> m=video 17834 RTP/AVP 115
> a=rtpmap:115 H263-1998/90000
> 
> I've tried doing things on the soft phone - I can increase the send bit rate
> and reduce the definition but its still choppy.
> 
> Is this some way to increase the bit-rate on freeswitch side?  
> 
> Owen



From Prometheus001 at gmx.net  Fri Mar 26 07:36:29 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 26 Mar 2010 15:36:29 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <4BA7943D.9030001@gmx.net>
References: <4B9F71AD.7070008@gmx.net>	<2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>
	<4BA7943D.9030001@gmx.net>
Message-ID: <4BACC66D.1090301@gmx.net>

Nobody has an idea how to force the 486 SIP message?

Best regards
Peter

Peter P GMX schrieb am 22.03.2010:
> I tried this, but only with half success
>
> after the phones responds with "do not disturb"
> (Variables from Info application
> variable_proto_specific_hangup_cause: [sip:480]
> variable_sip_hangup_phrase: [Do Not Disturb])
>
> I do process the following:
> <action application="bridge"
> data="loopback/hangup_causes_${sip_hangup_phrase}"/>
>
> In the following dialplan I process:
> <extension name="hangup-causes">
> <action application="info" data=""/>
> <condition field="destination_number" expression="hangup_causes_Do Not
> Disturb">
> <action application="sleep" data="1000"/>
> <action application="set" data="sip_ignore_remote_cause=true"/>
> <!-- <action application="respond" data="486"/> -->
> <action application="hangup" data="USER_BUSY"/>
> </condition>
> </extension>
> Freeswitch sends a 480 Message with
> Reason: Q.850;cause=17;text="USER_BUSY"
>
> This is a first step, but I need a 486 message as the PSTN still reponds
> "temporary unavailable" instead of "busy". Also "respond" with "486"
> does not work, as commented in the dialplan.
>
> It seems that the 480 message cannot be overriden in this case.
> Freeswitch insists on using 480
> mod_sofia.c:352 sofia/internal/203 at mydomain Overriding SIP cause 486
> with 480 from the other leg
>
> Any other idea to achieve a 486 message?
>
> Best regards
> Peter
>
> Leon de Rooij schrieb:
>   
>> Hi,
>>
>> Don't know if it helps, but the inline attribute can only be set on  
>> actions, not on conditions.
>>
>> Also, the hangup application is not allowed to be run inline:
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_dptools/mod_dptools.c?r=HEAD#l3047
>> (it has no SAF_ROUTING_EXEC flag)
>>
>> Perhaps you can try to transfer to another context directly after your  
>> bridge ? (where that second context contains an extension with  
>> conditions matching on variable_originate_disposition)
>>
>> kind regards,
>>
>> Leon
>>
>>
>> On Mar 16, 2010, at 12:55 PM, Peter P GMX wrote:
>>
>>   
>>     
>>> I have the folowing scenario:
>>>  A Snom phone has activated DND (do not disturb)
>>>  When the phone is called, it returns "480 Do Not Disturb" and then
>>> "404 Not Found".
>>>  Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
>>>  The PSTN then (wrongly) announces "this number is not listed".
>>>
>>> My question: How can we overcome this and announce back a different
>>> message to PSTN, but only on NO_USER_RESPONSE.
>>> I tried to continue on bridge and then this after the bridge command  
>>> the
>>> following:
>>>      <condition field="variable_originate_disposition"
>>> expression="NO_USER_RESPONSE" inline="true">
>>>        <action application="hangup" data="USER_BUSY"/>
>>>      </condition>
>>>
>>> But the installed Freeswith does not support the "inline" yet and we
>>> will not be able to update Freeswitch soon.
>>>
>>> So my question:
>>> Is there any other way to achieve this?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>     
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>   
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From brian at freeswitch.org  Fri Mar 26 07:45:38 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 26 Mar 2010 09:45:38 -0500
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <4BACC66D.1090301@gmx.net>
References: <4B9F71AD.7070008@gmx.net>	<2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>
	<4BA7943D.9030001@gmx.net> <4BACC66D.1090301@gmx.net>
Message-ID: <4EBF218D-331E-47FF-ACCD-FCDAAA94BB2D@freeswitch.org>

respond 486 or hangup USER_BUSY

/b

On Mar 26, 2010, at 9:36 AM, Peter P GMX wrote:

> Nobody has an idea how to force the 486 SIP message?
> 
> Best regards
> Peter



From Prometheus001 at gmx.net  Fri Mar 26 07:58:52 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 26 Mar 2010 15:58:52 +0100
Subject: [Freeswitch-users] ClueCon MMX - Call For Speakers! -we would
 like to speak
In-Reply-To: <87f2f3b91002030628n3b3bf512x68947758cf042ea@mail.gmail.com>
References: <87f2f3b91002030628n3b3bf512x68947758cf042ea@mail.gmail.com>
Message-ID: <4BACCBAC.6030402@gmx.net>

Hello Michael,

as last year we would like to attend (to speak) again on this
conference. Which information do you need from us for this event?

Best regards
Peter Steinbach

Michael Collins schrieb:
> Hello everyone!
>
> We are gearing up for ClueCon 2010 in August later this year. We are
> making the necessary arrangements for the conference facilities and
> rooming. Things are beginning to fall into place. Now we need to hear
> from you.
>
> We would like to put out a call for speakers for this year's event.
> Please contact us if you or your organization would like to give a
> presentation at ClueCon this year. We want to get the speakers
> scheduled as early as possible. Keep in mind that those organizations
> which sponsor ClueCon will be given the highest priority when it comes
> to scheduling. Please contact Brian West to discuss sponsorship
> opportunities for this year's event.
>
> We also would like to hear from the conference attendees: what would
> you like to see this year? Please give us your input. ClueCon is, of
> course, "By Developers, For Developers." However, developers come in
> all shapes and sizes and we would like have something for everyone.
> Please tell us what would make ClueCon MMX the best conference of the
> year!
>
> Stay tuned for more announcements. We look forward to hearing from you
> and seeing everyone this August in Chicago.
>
> -ClueCon team
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From erkan at speedingtrade.com  Thu Mar 25 23:45:41 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Fri, 26 Mar 2010 08:45:41 +0200
Subject: [Freeswitch-users] Set the outgoing caller id
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>

Hi all,

 

I worked since 2 years with asterisk. But now I must use the Windows Platform.

So I found Freeswitch.

The first works with Freeswitch are very good, but I can't set the outgoing caller id.

 

The outgoing caller id is shown at the phone on the end of the line.

 

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

 

....

<include>

  <context name="default">

 

   <extension name="voicetrading.com">

     <condition field="destination_number" expression="^00(\d+)$">

         <action application="set" data="effective_caller_id_name=009012345678"/>

         <action application="set" data="effective_caller_id_number=009012345678"/>

         <action application="set" data="outbound_caller_id_name=009012345678"/>

         <action application="set" data="outbound_caller_id_number=009012345678"/>

         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

     </condition>

   </extension>

....

 

Can anyone give me a idea for what can I do.

 

Kind regards

Erkan Unlu

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From david.ponzone at ipeva.fr  Fri Mar 26 08:09:03 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 26 Mar 2010 16:09:03 +0100
Subject: [Freeswitch-users] How to answer USER_BUSY on NO_USER_RESPONSE?
In-Reply-To: <4BACC66D.1090301@gmx.net>
References: <4B9F71AD.7070008@gmx.net>	<2B8BE348-65E6-4A32-8346-0BECF49323A4@scarlet-internet.nl>
	<4BA7943D.9030001@gmx.net> <4BACC66D.1090301@gmx.net>
Message-ID: <FCB09CC7-F09E-4071-89C1-2DC53DE24BC1@ipeva.fr>

Peter,

I had a similar issue and I did the following in my dialplan:

     <extension name="from_provider_to_internal_switch">
       <condition field="destination_number" expression="^(\d{10})$">
	<action application="set" data="hangup_after_bridge=true"/>
	<action application="set" data="continue_on_fail=21"/>
         <action application="bridge" data="{sip_cid_type=none}sofia/ 
gateway/otherswitch/$1"/>
         <action application="transfer" data="603to486"/>
       </condition>
     </extension>

     <extension name="603to486">
       <condition field="${proto_specific_hangup_cause}" expression="sip:603 
">
	<action application="set" data="sip_ignore_remote_cause=true"/>
         <action application="respond" data="486"/>
         <action application="hangup" data="USER_BUSY"/>
       </condition>
     </extension>

You would have to replace 21 by what the q850 code FS gets from the  
phone, or true, and 603 by 480.

It's probably not the best way to do it. A LUA script would be nicer.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 26/03/2010 ? 15:36, Peter P GMX a ?crit :

> Nobody has an idea how to force the 486 SIP message?
>
> Best regards
> Peter
>
> Peter P GMX schrieb am 22.03.2010:
>> I tried this, but only with half success
>>
>> after the phones responds with "do not disturb"
>> (Variables from Info application
>> variable_proto_specific_hangup_cause: [sip:480]
>> variable_sip_hangup_phrase: [Do Not Disturb])
>>
>> I do process the following:
>> <action application="bridge"
>> data="loopback/hangup_causes_${sip_hangup_phrase}"/>
>>
>> In the following dialplan I process:
>> <extension name="hangup-causes">
>> <action application="info" data=""/>
>> <condition field="destination_number" expression="hangup_causes_Do  
>> Not
>> Disturb">
>> <action application="sleep" data="1000"/>
>> <action application="set" data="sip_ignore_remote_cause=true"/>
>> <!-- <action application="respond" data="486"/> -->
>> <action application="hangup" data="USER_BUSY"/>
>> </condition>
>> </extension>
>> Freeswitch sends a 480 Message with
>> Reason: Q.850;cause=17;text="USER_BUSY"
>>
>> This is a first step, but I need a 486 message as the PSTN still  
>> reponds
>> "temporary unavailable" instead of "busy". Also "respond" with "486"
>> does not work, as commented in the dialplan.
>>
>> It seems that the 480 message cannot be overriden in this case.
>> Freeswitch insists on using 480
>> mod_sofia.c:352 sofia/internal/203 at mydomain Overriding SIP cause 486
>> with 480 from the other leg
>>
>> Any other idea to achieve a 486 message?
>>
>> Best regards
>> Peter
>>
>> Leon de Rooij schrieb:
>>
>>> Hi,
>>>
>>> Don't know if it helps, but the inline attribute can only be set on
>>> actions, not on conditions.
>>>
>>> Also, the hangup application is not allowed to be run inline:
>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/applications/mod_dptools/mod_dptools.c?r=HEAD#l3047
>>> (it has no SAF_ROUTING_EXEC flag)
>>>
>>> Perhaps you can try to transfer to another context directly after  
>>> your
>>> bridge ? (where that second context contains an extension with
>>> conditions matching on variable_originate_disposition)
>>>
>>> kind regards,
>>>
>>> Leon
>>>
>>>
>>> On Mar 16, 2010, at 12:55 PM, Peter P GMX wrote:
>>>
>>>
>>>
>>>> I have the folowing scenario:
>>>> A Snom phone has activated DND (do not disturb)
>>>> When the phone is called, it returns "480 Do Not Disturb" and then
>>>> "404 Not Found".
>>>> Freeswitch then announces NO_USER_RESPONSE to the caller (PSTN).
>>>> The PSTN then (wrongly) announces "this number is not listed".
>>>>
>>>> My question: How can we overcome this and announce back a different
>>>> message to PSTN, but only on NO_USER_RESPONSE.
>>>> I tried to continue on bridge and then this after the bridge  
>>>> command
>>>> the
>>>> following:
>>>>     <condition field="variable_originate_disposition"
>>>> expression="NO_USER_RESPONSE" inline="true">
>>>>       <action application="hangup" data="USER_BUSY"/>
>>>>     </condition>
>>>>
>>>> But the installed Freeswith does not support the "inline" yet and  
>>>> we
>>>> will not be able to update Freeswitch soon.
>>>>
>>>> So my question:
>>>> Is there any other way to achieve this?
>>>>
>>>> Best regards
>>>> Peter
>>>>

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From andrew at hijacked.us  Fri Mar 26 08:19:32 2010
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 26 Mar 2010 11:19:32 -0400
Subject: [Freeswitch-users] Set the outgoing caller id
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
Message-ID: <20100326151932.GE1969@hijacked.us>

On Fri, Mar 26, 2010 at 08:45:41AM +0200, Erkan ?nl? wrote:
 
> Can anyone give me a idea for what can I do.
> 

Try the origination_caller_id variables.

Andrew


From msc at freeswitch.org  Fri Mar 26 08:22:27 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 08:22:27 -0700
Subject: [Freeswitch-users] How To: Sip Logging and Freeswitch
In-Reply-To: <1269597246475-4802944.post@n2.nabble.com>
References: <1269597246475-4802944.post@n2.nabble.com>
Message-ID: <87f2f3b91003260822rf955317t80ccc50f390cddb7@mail.gmail.com>

This looks like something that could go on the wiki. How about creating a
new page and linking to it from this page:
http://wiki.freeswitch.org/wiki/Examples

Be sure to give it a category of "Examples". If you need any help at all
with wiki markup or getting started just let me know. Join us on IRC:
#freeswitch at irc.freenode.net. My nick is mercutioviz.

Thanks for sharing!
-MC

On Fri, Mar 26, 2010 at 2:54 AM, OwenW <williams.owen at gmail.com> wrote:

>
> This is a quick guide as to one approach to logging the sip messages
> between
> a Sip phone and freeswitch.  There are other methods (e.g. wireshark) -
> feel
> free to reply if you have other alternatives.
>
> It involves putting a java TCP Proxy (grinder) in front of freeswitch and
> reconfiguring your SipPhone to send SIP to the proxy.
>
> The software (grinder) is Java based so will require an JDK.
>
> Once you've downloaded Grinder create a file called startProxy5061.cmd and
> put the following into it:
> call setGrinderEnv.cmd
> java -cp %CLASSPATH% net.grinder.TCPProxy -console -localhost 192.168.30.10
> -localport 5061 -remotehost 192.168.30.10 -remoteport 5060 >
> freeswitch_sip.log
>
> The proxy forwards all traffic it receives on 5061 to 5060 and logs to
> freeswitch_sip.log.
>
> You'll need to change the IPs above.
> Create another file called setGrinderEnv.cmd and put the following in to
> it:
> set GRINDERPATH=C:\dev\apps\grinder-3.3
> set GRINDERPROPERTIES=C:\dev\apps\grinder-3.3\grinder.properties
> set CLASSPATH=%GRINDERPATH%\lib\grinder.jar;%CLASSPATH%
> set JAVA_HOME=c:\Program Files\java\jdk1.6.0_11
> PATH=%JAVA_HOME%\bin;%PATH%
>
> You'll need to change the paths above.
>
> The soft phone will need to be configured to use port 5061 instead of 5060
> as the proxy.
>
> Owen
>
>
>
> --
> View this message in context:
> http://n2.nabble.com/How-To-Sip-Logging-and-Freeswitch-tp4802944p4802944.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Mar 26 08:26:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 08:26:09 -0700
Subject: [Freeswitch-users] ClueCon MMX - Call For Speakers! -we would
	like to speak
In-Reply-To: <4BACCBAC.6030402@gmx.net>
References: <87f2f3b91002030628n3b3bf512x68947758cf042ea@mail.gmail.com>
	<4BACCBAC.6030402@gmx.net>
Message-ID: <87f2f3b91003260826l637d516l662d8ed67f1c00c3@mail.gmail.com>

On Fri, Mar 26, 2010 at 7:58 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Michael,
>
> as last year we would like to attend (to speak) again on this
> conference. Which information do you need from us for this event?
>
> Best regards
> Peter Steinbach
>
Thank you Peter! Please supply a working title or subject. We would also
like a brief abstract that we could put on the schedule on the cluecon
website. Also, we are trying something different this year: all talks will
be 30 minutes except for a few specific presentations.

Please let me know if you have any questions.
Thanks,
MC
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From kristian.kielhofner at gmail.com  Fri Mar 26 08:26:26 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Fri, 26 Mar 2010 11:26:26 -0400
Subject: [Freeswitch-users] Returning the ip address of a registered user
Message-ID: <4d15ff861003260826u499eaaa2v34d58ce6f2a9bc61@mail.gmail.com>

Hello everyone,

  What's the easiest/lightest way to return just the IP address and
port of a registered user?

  So far I can see using function_sofia_contact and parsing that (not
preferred because it won't detect NAT) OR executing the SQL lookups
from FS and returning the IP and port as known to FreeSWITCH.

  Any ideas?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Fri Mar 26 08:29:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 08:29:48 -0700
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
In-Reply-To: <FEFF4F1E-3493-448A-B94B-E95BC18058DC@freeswitch.org>
References: <87k4xlga1k.fsf@quasar.esben-stien.name>
	<87pr6yz5wa.fsf@quasar.esben-stien.name>
	<1269532290668-4798438.post@n2.nabble.com>
	<191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com>
	<1269596112846-4802883.post@n2.nabble.com>
	<3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org>
	<1269613532826-4804168.post@n2.nabble.com>
	<FEFF4F1E-3493-448A-B94B-E95BC18058DC@freeswitch.org>
Message-ID: <87f2f3b91003260829p195dcab9n6a17991af942b83d@mail.gmail.com>

What?! You mean FreeSWITCH doesn't have mod_change_crap_to_gold?! :P

Yep, this sounds like a classic case of garbage in, garbage out...
-MC

On Fri, Mar 26, 2010 at 7:32 AM, Brian West <brian at freeswitch.org> wrote:

> No because FreeSWITCH doesn't encode the video... its returning exactly
> what was received packet by packet.  So to clarify you sent crappy video to
> FreeSWITCH, It wrote it down, You called back and viewed the crappy video
> coming back at you.  :P
>
> /b
>
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From msc at freeswitch.org  Fri Mar 26 08:37:10 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 08:37:10 -0700
Subject: [Freeswitch-users] Set the outgoing caller id
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
Message-ID: <87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com>

Some providers have different ways of handling this. Using
"effective_caller_id_number" and name are the usual ways of doing it.
However some providers pick up the caller ID from a different field. Try
experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/
voicetrading.com/00000$1"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/
voicetrading.com/00000$1"/>

Please try it and let us know what happens.
-MC

2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

>  Hi all,
>
>
>
> I worked since 2 years with asterisk. But now I must use the Windows
> Platform.
>
> So I found Freeswitch.
>
> The first works with Freeswitch are very good, but I can?t set the outgoing
> caller id.
>
>
>
> The outgoing caller id is shown at the phone on the end of the line.
>
>
>
> I try some different variables but without result. My voip provider support
> to set the callerid, in asterisk it works.
>
>   <action application="bridge" data="sofia/gateway/
> voicetrading.com/00000$1"/>
>
> ?.
>
> <include>
>
>   <context name="default">
>
>
>
>    <extension name="voicetrading.com">
>
>      <condition field="destination_number" expression="^00(\d+)$">
>
>          <action application="set"
> data="effective_caller_id_name=009012345678"/>
>
>          <action application="set"
> data="effective_caller_id_number=009012345678"/>
>
>          <action application="set"
> data="outbound_caller_id_name=009012345678"/>
>
>          <action application="set"
> data="outbound_caller_id_number=009012345678"/>
>
>          <action application="bridge" data="sofia/gateway/
> voicetrading.com/00000$1"/>
>
>      </condition>
>
>    </extension>
>
> ?.
>
>
>
> Can anyone give me a idea for what can I do.
>
>
>
> Kind regards
>
> Erkan Unlu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stevendt at primrosebank.net  Fri Mar 26 08:51:43 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 26 Mar 2010 15:51:43 -0000
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>
	<d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com>
Message-ID: <E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>

Yehavi,

thanks a lot for the reply - not the answer that I wanted, but it does confirm my findings and tell me that there's no point in spending more time & effort on these phones. 

Before I went for the ST2030, I checked out the FreeSWITCH Interop Wiki and saw that the ST2030 was reported as working with FreeSWITCH, unfortunately, the Wiki does not give any information on how good (or bad) the phones actually are. I also did some "Googling" and the phone appeared to get good reports (from Asterisk users), so I thought I was on safe ground. Obviously not though.

Oh well, I'll keep trying to find a decent VOIP phone with full SpeakerPhone functionality that does not have the footprint of the Cisco 794x.

regards
Dave
  ----- Original Message ----- 
  From: Yehavi Bourvine 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 26, 2010 4:59 AM
  Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones


  We have one ST2030 and one other model of them which we bought for evaluation. We have the same problem as you describe, both with Freeswitch and Asterisk. That's why we have only one unit and not more...

  from my experience, you get what you pay for.

                    __Yehavi:


  2010/3/25 Dave Stevenson <stevendt at primrosebank.net>
    Hi,

    is anyone using Thomson ST2030 phones with FreeSwitch ?

    I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite a bit of noise (hiss) on the line and some breakup of the voice. The phones are all on a private LAN behind a router - all external calls are over the PSTN - no external IP calls are made (I don't use a VOIP provider).

    I have loaded the latest Thomson SIP firmware (2.69). There are a couple of options on the phone that I've tried such as Silence Suppression, Acoustic Echo Cancellation and Packet Loss Compensation with no improvement. Most of the other options are beyond me I'm afraid, so I'd be really interested to hear from anyone who has good (or even acceptable !) results from the ST2030 - particularly if they could share their phone configuration

    regards
    Dave

    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From david.ponzone at gmail.com  Fri Mar 26 09:00:56 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 26 Mar 2010 17:00:56 +0100
Subject: [Freeswitch-users] Returning the ip address of a registered user
In-Reply-To: <4d15ff861003260826u499eaaa2v34d58ce6f2a9bc61@mail.gmail.com>
References: <4d15ff861003260826u499eaaa2v34d58ce6f2a9bc61@mail.gmail.com>
Message-ID: <5A4DEE1C-8BE0-42C7-9841-35A310327E72@gmail.com>

Kristian,

with sofia_contact, you should get a string which contains fs_path=sip 
%3Auserid%40NAT_IP%3ANAT_PORT%3B.....
where NAT_IP and NAT_PORT are the values you are looking for.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 26/03/2010 ? 16:26, Kristian Kielhofner a ?crit :

> Hello everyone,
>
>  What's the easiest/lightest way to return just the IP address and
> port of a registered user?
>
>  So far I can see using function_sofia_contact and parsing that (not
> preferred because it won't detect NAT) OR executing the SQL lookups
> from FS and returning the IP and port as known to FreeSWITCH.
>
>  Any ideas?
>
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From yehavi.bourvine at gmail.com  Fri Mar 26 09:04:59 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 26 Mar 2010 18:04:59 +0200
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
In-Reply-To: <E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>
	<d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com>
	<E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
Message-ID: <d361ca7b1003260904o25545cdfod53abbcf931de43e@mail.gmail.com>

There is one thing which might help you, but since I am at home I don't
remember the exact details. There is some codec parameter which should be
set  (on the phone) to 20mSec instead of 30mSec. I don't remember which one,
but look on LinkSys documents on Freeswitch wiki - it has the same problem.

                       Regards, __Yehavi:

 2010/3/26 Dave Stevenson <stevendt at primrosebank.net>

>  Yehavi,
>
> thanks a lot for the reply - not the answer that I wanted, but it does
> confirm my findings and tell me that there's no point in spending more time
> & effort on these phones.
>
> Before I went for the ST2030, I checked out the FreeSWITCH Interop Wiki and
> saw that the ST2030 was reported as working with FreeSWITCH, unfortunately,
> the Wiki does not give any information on how good (or bad) the phones
> actually are. I also did some "Googling" and the phone appeared to get good
> reports (from Asterisk users), so I thought I was on safe ground. Obviously
> not though.
>
> Oh well, I'll keep trying to find a decent VOIP phone with full
> SpeakerPhone functionality that does not have the footprint of the Cisco
> 794x.
>
> regards
> Dave
>
>   ----- Original Message -----
> *From:* Yehavi Bourvine <yehavi.bourvine at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, March 26, 2010 4:59 AM
> *Subject:* Re: [Freeswitch-users] Thomson ST2030 IP Phones
>
>  We have one ST2030 and one other model of them which we bought for
> evaluation. We have the same problem as you describe, both with Freeswitch
> and Asterisk. That's why we have only one unit and not more...
>
> from my experience, you get what you pay for.
>
>                   __Yehavi:
>
>  2010/3/25 Dave Stevenson <stevendt at primrosebank.net>
>
>>  Hi,
>>
>> is anyone using Thomson ST2030 phones with FreeSwitch ?
>>
>> I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the
>> Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite
>> a bit of noise (hiss) on the line and some breakup of the voice. The phones
>> are all on a private LAN behind a router - all external calls are over the
>> PSTN - no external IP calls are made (I don't use a VOIP provider).
>>
>> I have loaded the latest Thomson SIP firmware (2.69). There are a couple
>> of options on the phone that I've tried such as Silence Suppression,
>> Acoustic Echo Cancellation and Packet Loss Compensation with no improvement.
>> Most of the other options are beyond me I'm afraid, so I'd be really
>> interested to hear from anyone who has good (or even acceptable !) results
>> from the ST2030 - particularly if they could share their phone configuration
>>
>> regards
>> Dave
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Fri Mar 26 09:08:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 26 Mar 2010 17:08:16 +0100
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
In-Reply-To: <E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com>
	<d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com>
	<E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
Message-ID: <2F615A16-AAAC-4F4E-BA5E-82B8E839E2F1@gmail.com>

Dave,

We have quite a few ST2030 deployed, but they run MGCP and they are  
connected on a commercial IP-Centrex platform.

I am not going to say that's the best phone in the world, but we never  
had the issues you're describing (and the SIP firmware is considered  
more advanced than the MGCP one).
And we are using G729.

So either the G711 code in the phone is crap (I have to admit I never  
used it), or something is wrong somewhere.
I would recommend you disable silence suppression (that's crap  
anyway), and then take a RTP capture of the call, before and after FS.
Then decode the RTP streams with wireshark, and listen to it.
The point is to check if the issue is in the RTP stream sent by the  
phone, or if it appears after FS (unlikely), or if it is because of  
the way the other phone decodes it.
Are you having the issue from Thomson to Thomson ?
What about if you enable bypass_media ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 26/03/2010 ? 16:51, Dave Stevenson a ?crit :

> Yehavi,
>
> thanks a lot for the reply - not the answer that I wanted, but it  
> does confirm my findings and tell me that there's no point in  
> spending more time & effort on these phones.
>
> Before I went for the ST2030, I checked out the FreeSWITCH Interop  
> Wiki and saw that the ST2030 was reported as working with  
> FreeSWITCH, unfortunately, the Wiki does not give any information on  
> how good (or bad) the phones actually are. I also did some  
> "Googling" and the phone appeared to get good reports (from Asterisk  
> users), so I thought I was on safe ground. Obviously not though.
>
> Oh well, I'll keep trying to find a decent VOIP phone with full  
> SpeakerPhone functionality that does not have the footprint of the  
> Cisco 794x.
>
> regards
> Dave
> ----- Original Message -----
> From: Yehavi Bourvine
> To: freeswitch-users at lists.freeswitch.org
> Sent: Friday, March 26, 2010 4:59 AM
> Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones
>
> We have one ST2030 and one other model of them which we bought for  
> evaluation. We have the same problem as you describe, both with  
> Freeswitch and Asterisk. That's why we have only one unit and not  
> more...
>
> from my experience, you get what you pay for.
>
>                   __Yehavi:
>
> 2010/3/25 Dave Stevenson <stevendt at primrosebank.net>
> Hi,
>
> is anyone using Thomson ST2030 phones with FreeSwitch ?
>
> I have a mixture of Cisco 7940 and the ST2030s. The voice quality of  
> the Cisco phones seems fine, but the Thomson phones are pretty poor.  
> I get quite a bit of noise (hiss) on the line and some breakup of  
> the voice. The phones are all on a private LAN behind a router - all  
> external calls are over the PSTN - no external IP calls are made (I  
> don't use a VOIP provider).
>
> I have loaded the latest Thomson SIP firmware (2.69). There are a  
> couple of options on the phone that I've tried such as Silence  
> Suppression, Acoustic Echo Cancellation and Packet Loss Compensation  
> with no improvement. Most of the other options are beyond me I'm  
> afraid, so I'd be really interested to hear from anyone who has good  
> (or even acceptable !) results from the ST2030 - particularly if  
> they could share their phone configuration
>
> regards
> Dave
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Fri Mar 26 09:11:51 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 26 Mar 2010 12:11:51 -0400
Subject: [Freeswitch-users] Freeswitch Video Capture and Playback
References: <87k4xlga1k.fsf@quasar.esben-stien.name><87pr6yz5wa.fsf@quasar.esben-stien.name><1269532290668-4798438.post@n2.nabble.com><191c3a031003251208k1e662da8tfa8001298ce6bfa7@mail.gmail.com><1269596112846-4802883.post@n2.nabble.com><3bf3ede237d26c7b3a272797855d2fb1@thom.fr.eu.org><1269613532826-4804168.post@n2.nabble.com><FEFF4F1E-3493-448A-B94B-E95BC18058DC@freeswitch.org>
	<87f2f3b91003260829p195dcab9n6a17991af942b83d@mail.gmail.com>
Message-ID: <08A88962B9854922949B1F283FD2D9E5@MOBILEE1705>

the day when FS will encode videos there will be 16 nano CPUS at 8ghz each on standard servers....
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 26, 2010 11:29 AM
  Subject: Re: [Freeswitch-users] Freeswitch Video Capture and Playback


  What?! You mean FreeSWITCH doesn't have mod_change_crap_to_gold?! :P

  Yep, this sounds like a classic case of garbage in, garbage out...
  -MC


  On Fri, Mar 26, 2010 at 7:32 AM, Brian West <brian at freeswitch.org> wrote:

    No because FreeSWITCH doesn't encode the video... its returning exactly what was received packet by packet.  So to clarify you sent crappy video to FreeSWITCH, It wrote it down, You called back and viewed the crappy video coming back at you.  :P

    /b



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From stevendt at primrosebank.net  Fri Mar 26 09:42:48 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 26 Mar 2010 16:42:48 -0000
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com><d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com><E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
	<d361ca7b1003260904o25545cdfod53abbcf931de43e@mail.gmail.com>
Message-ID: <FA08733B626C4990B1EB91AD6637D2CD@bp1.ad.bp.com>

Thanks Yehavi,

I did need to change the codec "packetization" parameters (there's one for each codec) from the defaults to 20ms which helped one problem but did not cure the rest of the quality issues

regards
Dave
  ----- Original Message ----- 
  From: Yehavi Bourvine 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 26, 2010 4:04 PM
  Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones


  There is one thing which might help you, but since I am at home I don't remember the exact details. There is some codec parameter which should be set  (on the phone) to 20mSec instead of 30mSec. I don't remember which one, but look on LinkSys documents on Freeswitch wiki - it has the same problem.

                         Regards, __Yehavi:


  2010/3/26 Dave Stevenson <stevendt at primrosebank.net>
    Yehavi,

    thanks a lot for the reply - not the answer that I wanted, but it does confirm my findings and tell me that there's no point in spending more time & effort on these phones. 

    Before I went for the ST2030, I checked out the FreeSWITCH Interop Wiki and saw that the ST2030 was reported as working with FreeSWITCH, unfortunately, the Wiki does not give any information on how good (or bad) the phones actually are. I also did some "Googling" and the phone appeared to get good reports (from Asterisk users), so I thought I was on safe ground. Obviously not though.

    Oh well, I'll keep trying to find a decent VOIP phone with full SpeakerPhone functionality that does not have the footprint of the Cisco 794x.

    regards
    Dave
      ----- Original Message ----- 
      From: Yehavi Bourvine 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, March 26, 2010 4:59 AM
      Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones


      We have one ST2030 and one other model of them which we bought for evaluation. We have the same problem as you describe, both with Freeswitch and Asterisk. That's why we have only one unit and not more...

      from my experience, you get what you pay for.

                        __Yehavi:


      2010/3/25 Dave Stevenson <stevendt at primrosebank.net>
        Hi,

        is anyone using Thomson ST2030 phones with FreeSwitch ?

        I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite a bit of noise (hiss) on the line and some breakup of the voice. The phones are all on a private LAN behind a router - all external calls are over the PSTN - no external IP calls are made (I don't use a VOIP provider).

        I have loaded the latest Thomson SIP firmware (2.69). There are a couple of options on the phone that I've tried such as Silence Suppression, Acoustic Echo Cancellation and Packet Loss Compensation with no improvement. Most of the other options are beyond me I'm afraid, so I'd be really interested to hear from anyone who has good (or even acceptable !) results from the ST2030 - particularly if they could share their phone configuration

        regards
        Dave

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From stevendt at primrosebank.net  Fri Mar 26 10:04:37 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 26 Mar 2010 17:04:37 -0000
Subject: [Freeswitch-users] Thomson ST2030 IP Phones
References: <65939C5FD4EE46058614FBA48B9BD58D@bp1.ad.bp.com><d361ca7b1003252159l470763b7t38ad7a1643ef1e67@mail.gmail.com><E3E8F5AB81F344078B51F8E90E8F82C7@bp1.ad.bp.com>
	<2F615A16-AAAC-4F4E-BA5E-82B8E839E2F1@gmail.com>
Message-ID: <5DDBD6632C57477FA1F194F9A80CB84B@bp1.ad.bp.com>

David,

thanks for the info - maybe I'm giving up too soon !

I've not used WireShark before - I downloaded a copy a while ago, but it looked a bit beyond me, it sounds like this might be the driver for me to try to get to grips with it though.

As you suggest, On or Off - the Silence Suppression did not actually seem to do anything noticeable but it is currently off.

There does not seem to be much, if any, difference when calling Thomson-Thomson or Thomson-Cisco, but there does seem to be a marked difference between those combinations and when talking Cisco-Cisco.

I have not tried "bypass-media" - off to read up about it now,

regards
Dave
  ----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, March 26, 2010 4:08 PM
  Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones


  Dave,


  We have quite a few ST2030 deployed, but they run MGCP and they are connected on a commercial IP-Centrex platform.


  I am not going to say that's the best phone in the world, but we never had the issues you're describing (and the SIP firmware is considered more advanced than the MGCP one).
  And we are using G729.


  So either the G711 code in the phone is crap (I have to admit I never used it), or something is wrong somewhere.
  I would recommend you disable silence suppression (that's crap anyway), and then take a RTP capture of the call, before and after FS.
  Then decode the RTP streams with wireshark, and listen to it.
  The point is to check if the issue is in the RTP stream sent by the phone, or if it appears after FS (unlikely), or if it is because of the way the other phone decodes it.
  Are you having the issue from Thomson to Thomson ?
  What about if you enable bypass_media ?


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 26/03/2010 ? 16:51, Dave Stevenson a ?crit :


    Yehavi,

    thanks a lot for the reply - not the answer that I wanted, but it does confirm my findings and tell me that there's no point in spending more time & effort on these phones.

    Before I went for the ST2030, I checked out the FreeSWITCH Interop Wiki and saw that the ST2030 was reported as working with FreeSWITCH, unfortunately, the Wiki does not give any information on how good (or bad) the phones actually are. I also did some "Googling" and the phone appeared to get good reports (from Asterisk users), so I thought I was on safe ground. Obviously not though.

    Oh well, I'll keep trying to find a decent VOIP phone with full SpeakerPhone functionality that does not have the footprint of the Cisco 794x.

    regards
    Dave
      ----- Original Message -----
      From: Yehavi Bourvine
      To: freeswitch-users at lists.freeswitch.org
      Sent: Friday, March 26, 2010 4:59 AM
      Subject: Re: [Freeswitch-users] Thomson ST2030 IP Phones


      We have one ST2030 and one other model of them which we bought for evaluation. We have the same problem as you describe, both with Freeswitch and Asterisk. That's why we have only one unit and not more...

      from my experience, you get what you pay for.

                        __Yehavi:


      2010/3/25 Dave Stevenson <stevendt at primrosebank.net>
        Hi,

        is anyone using Thomson ST2030 phones with FreeSwitch ?

        I have a mixture of Cisco 7940 and the ST2030s. The voice quality of the Cisco phones seems fine, but the Thomson phones are pretty poor. I get quite a bit of noise (hiss) on the line and some breakup of the voice. The phones are all on a private LAN behind a router - all external calls are over the PSTN - no external IP calls are made (I don't use a VOIP provider).

        I have loaded the latest Thomson SIP firmware (2.69). There are a couple of options on the phone that I've tried such as Silence Suppression, Acoustic Echo Cancellation and Packet Loss Compensation with no improvement. Most of the other options are beyond me I'm afraid, so I'd be really interested to hear from anyone who has good (or even acceptable !) results from the ST2030 - particularly if they could share their phone configuration

        regards
        Dave

        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        http://www.freeswitch.org







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From anthony.minessale at gmail.com  Fri Mar 26 10:39:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 12:39:57 -0500
Subject: [Freeswitch-users] Fwd: Sofia crashing when registering
In-Reply-To: <64CC5F22-1D5A-4441-8AFD-24D0228F0CB6@5ninesolutions.com>
References: <E45C5A92-C8D0-47A0-BA11-376C12DBF0C0@5ninesolutions.com>
	<5BFF727B-6A2C-457C-950D-ACA3565F43B8@5ninesolutions.com>
	<8585B8A8-BE67-4DD6-83CE-0B572E9B5750@freeswitch.org>
	<D4E4B02A-B3A4-49F6-AC8E-676427A539EA@5ninesolutions.com>
	<016AF215-E630-454B-90E1-DA13A5A006A8@freeswitch.org>
	<03BD9C42-C634-4020-ADD0-981531DB3B12@5ninesolutions.com>
	<b39273531003250413p70389210ya0ebe797d273065d@mail.gmail.com>
	<191c3a031003250626h74092581g14aa4185e69d02ca@mail.gmail.com>
	<191c3a031003250627r71d5a731p447b207f51fe96eb@mail.gmail.com>
	<64CC5F22-1D5A-4441-8AFD-24D0228F0CB6@5ninesolutions.com>
Message-ID: <191c3a031003261039w73dbc689v2f20c3803a03dd0f@mail.gmail.com>

I found the problem in the sofia-sip library, there was a typo causing your
endless loop in a very rare condition triggered by your provider's proxy.

They are using an unsafe version of SIP reg BTW, that is vulnerable to a
replay attack.

Its fixed in r17114



On Thu, Mar 25, 2010 at 12:16 PM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Sure, I don't mind doing the legwork.  Logging now.  Again I apologize for
> the issues before.
>
> Spencer
>
> On Mar 25, 2010, at 6:27 AM, Anthony Minessale wrote:
>
> since I forgot to mention,
>
> restart FS or at leasr mod_logfile after making the changes to the config
>
>
> On Thu, Mar 25, 2010 at 8:26 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> If you recall, I tried to help you with this already once.
>> I spent a long time on your box building out fresh trunk and configuring
>> it to trace the problem.
>> Then you told me you erased it all by accident =/
>>
>> I suppose we can try again but I would appreciate if if you gather the
>> info for me.
>>
>> edit conf/autoload_configs/logfile.conf.xml
>>
>> look for this line:
>>
>> <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>>
>> replace it with:
>>
>> <map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>
>>
>>
>> in the cli:
>>
>> sofia profile internal siptrace on
>> sofia loglevel all 9
>>
>> (repeat for any other profiles)
>>
>> also run wireshark or tcpdump to capture all traffic to that particular
>> provider
>>
>> When it happens, stop FS and the capture and get me access to the
>> resulting files.
>>
>> This is a rare issue triggered by some strange sip message.
>> What we are trying to do here is get a copy of the traffic that triggers
>> it so we can stop it.
>>
>>
>>
>>
>> On Thu, Mar 25, 2010 at 6:13 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>>> Next time it happens, prior to a restart of the freeswitch process, run
>>> the fscore_pb script to get us a set of backtraces of the running process.
>>>
>>> support-d/fscore_pb gcore <yourname>
>>>
>>> try running it before hand to ensure you get expected results.  Getting a
>>> core does pause the process for a short period of time, so do your test when
>>> someone won't notice a RTP hickup.
>>>
>>> On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <
>>> spencer at 5ninesolutions.com> wrote:
>>>
>>>> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes
>>>> (1-n) -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is doing is
>>>> bridging calls from Asterisk to the ITSP.  All media is routed point to
>>>> point.  The dialplan is very simple as you can imagine.  If I disable
>>>> registrations to Bandtel everything seems to be ok.  but to my eye I don't
>>>> see anything weird about the format of the register that might cause a
>>>> problem.
>>>>
>>>> Spencer
>>>>
>>>> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>>>>
>>>> Odd nobody else has reported this behavior.  Have you viewed the wiki
>>>> pages on this topic of running under a vm?
>>>> /b
>>>>
>>>> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>>>>
>>>> I dunno, just thought I'd try something different.. :-)
>>>>
>>>> root at sip:/usr/local/freeswitch/bin# uname -r
>>>> 2.6.31-20-generic
>>>>
>>>> I'd much rather use CentOS.  The Ubuntu machine is running under Xen
>>>> 3.4.1 if that matters.
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From mike at jerris.com  Fri Mar 26 10:59:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Mar 2010 13:59:17 -0400
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap committors.
	ACTION REQUIRED
Message-ID: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>

We are in process of our migration from svn to git for our main source control.  We will likely still have a read only svn mirror, but all future commits will go to git after the migration.  In order to proceed with this process, I will need your full name, public key, email address, and svn username.  If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo.

For information on how to use git: http://git-scm.com/documentation , http://git-scm.com/course/svn.html
For information on how to generate a public key if you do not already have one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html

For general users, more information will follow on these changes.  For now, nothing has changed.  Around April 1, there will likely be a 1 time need to do a full checkout if you plan on continuing to use the svn mirror, or a fresh git checkout if you plan on switching to git.

If you have any other questions about the process or changes, feel free to ask.

Mike



From spencer at 5ninesolutions.com  Fri Mar 26 11:01:48 2010
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Fri, 26 Mar 2010 11:01:48 -0700
Subject: [Freeswitch-users] Fwd: Sofia crashing when registering
Message-ID: <191c3a031003261039w73dbc689v2f20c3803a03dd0f@mail.gmail.com>

Thanks!

Anthony Minessale <anthony.minessale at gmail.com> wrote:

>I found the problem in the sofia-sip library, there was a typo causing your
>endless loop in a very rare condition triggered by your provider's proxy.
>
>They are using an unsafe version of SIP reg BTW, that is vulnerable to a
>replay attack.
>
>Its fixed in r17114
>
>
>
>On Thu, Mar 25, 2010 at 12:16 PM, Spencer Thomason <
>spencer at 5ninesolutions.com> wrote:
>
>> Sure, I don't mind doing the legwork.  Logging now.  Again I apologize for
>> the issues before.
>>
>> Spencer
>>
>> On Mar 25, 2010, at 6:27 AM, Anthony Minessale wrote:
>>
>> since I forgot to mention,
>>
>> restart FS or at leasr mod_logfile after making the changes to the config
>>
>>
>> On Thu, Mar 25, 2010 at 8:26 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> If you recall, I tried to help you with this already once.
>>> I spent a long time on your box building out fresh trunk and configuring
>>> it to trace the problem.
>>> Then you told me you erased it all by accident =/
>>>
>>> I suppose we can try again but I would appreciate if if you gather the
>>> info for me.
>>>
>>> edit conf/autoload_configs/logfile.conf.xml
>>>
>>> look for this line:
>>>
>>> <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>>>
>>> replace it with:
>>>
>>> <map name="all" value="debug,info,notice,warning,err,crit,alert,console"/>
>>>
>>>
>>> in the cli:
>>>
>>> sofia profile internal siptrace on
>>> sofia loglevel all 9
>>>
>>> (repeat for any other profiles)
>>>
>>> also run wireshark or tcpdump to capture all traffic to that particular
>>> provider
>>>
>>> When it happens, stop FS and the capture and get me access to the
>>> resulting files.
>>>
>>> This is a rare issue triggered by some strange sip message.
>>> What we are trying to do here is get a copy of the traffic that triggers
>>> it so we can stop it.
>>>
>>>
>>>
>>>
>>> On Thu, Mar 25, 2010 at 6:13 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>>> Next time it happens, prior to a restart of the freeswitch process, run
>>>> the fscore_pb script to get us a set of backtraces of the running process.
>>>>
>>>> support-d/fscore_pb gcore <yourname>
>>>>
>>>> try running it before hand to ensure you get expected results.  Getting a
>>>> core does pause the process for a short period of time, so do your test when
>>>> someone won't notice a RTP hickup.
>>>>
>>>> On Thu, Mar 25, 2010 at 1:10 AM, Spencer Thomason <
>>>> spencer at 5ninesolutions.com> wrote:
>>>>
>>>>> Lucky me :-)  I have.  Its an extremely basic setup.  Asterisk boxes
>>>>> (1-n) -> Freeswitch (as an SBC) -> ITSP -> PSTN.  All Freeswitch is doing is
>>>>> bridging calls from Asterisk to the ITSP.  All media is routed point to
>>>>> point.  The dialplan is very simple as you can imagine.  If I disable
>>>>> registrations to Bandtel everything seems to be ok.  but to my eye I don't
>>>>> see anything weird about the format of the register that might cause a
>>>>> problem.
>>>>>
>>>>> Spencer
>>>>>
>>>>> On Mar 24, 2010, at 10:43 PM, Brian West wrote:
>>>>>
>>>>> Odd nobody else has reported this behavior.  Have you viewed the wiki
>>>>> pages on this topic of running under a vm?
>>>>> /b
>>>>>
>>>>> On Mar 25, 2010, at 12:37 AM, Spencer Thomason wrote:
>>>>>
>>>>> I dunno, just thought I'd try something different.. :-)
>>>>>
>>>>> root at sip:/usr/local/freeswitch/bin# uname -r
>>>>> 2.6.31-20-generic
>>>>>
>>>>> I'd much rather use CentOS.  The Ubuntu machine is running under Xen
>>>>> 3.4.1 if that matters.
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> -Rupa
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>-- 
>Anthony Minessale II
>
>FreeSWITCH http://www.freeswitch.org/
>ClueCon http://www.cluecon.com/
>Twitter: http://twitter.com/FreeSWITCH_wire
>
>AIM: anthm
>MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>IRC: irc.freenode.net #freeswitch
>
>FreeSWITCH Developer Conference
>sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>pstn:+19193869900
>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

From infos at madovsky.org  Fri Mar 26 11:11:16 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 26 Mar 2010 14:11:16 -0400
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
	committors.ACTION REQUIRED
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
Message-ID: <5008F84104E54F11AC95D38CD0BD3F32@MOBILEE1705>

for update with make current it will be transparent ?

----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-dev at lists.freeswitch.org>; "freeswitch-users" 
<freeswitch-users at lists.freeswitch.org>
Sent: Friday, March 26, 2010 1:59 PM
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap 
committors.ACTION REQUIRED


> We are in process of our migration from svn to git for our main source 
> control.  We will likely still have a read only svn mirror, but all future 
> commits will go to git after the migration.  In order to proceed with this 
> process, I will need your full name, public key, email address, and svn 
> username.  If you have outstanding svn branches that you would like to 
> keep, please contact me with these details as well, I do not currently 
> plan on merging any branches from svn to the new git repo.
>
> For information on how to use git: http://git-scm.com/documentation , 
> http://git-scm.com/course/svn.html
> For information on how to generate a public key if you do not already have 
> one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html
>
> For general users, more information will follow on these changes.  For 
> now, nothing has changed.  Around April 1, there will likely be a 1 time 
> need to do a full checkout if you plan on continuing to use the svn 
> mirror, or a fresh git checkout if you plan on switching to git.
>
> If you have any other questions about the process or changes, feel free to 
> ask.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From freeswitch-list at puzzled.xs4all.nl  Fri Mar 26 11:15:17 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Fri, 26 Mar 2010 19:15:17 +0100
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
Message-ID: <4BACF9B5.70803@puzzled.xs4all.nl>

On 03/26/2010 10:01 AM, Itamar Reis Peixoto wrote:
> On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon<gcd at i.ph>  wrote:
>> hello everybody,
>>
>> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
>> brand/model you had and your go/no-go recommendation.
>>
>> tks a lot,
>> nandy
>>
>
> portech ?

2N VoiceBlue. I only know they work fine with Asterisk. Iirc they are 
more expensive than the Portech boxes.

See http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway.html

Regards,
Patrick


From msc at freeswitch.org  Fri Mar 26 11:18:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 11:18:11 -0700
Subject: [Freeswitch-users] FreeSWITCH Pub Conf is bridged to the VUC
	conference
Message-ID: <87f2f3b91003261118h14541896j57b5602744f39088@mail.gmail.com>

Feel free to call in to the FS public conference to listen in as the FS team
and the VUC gang talk tech!

SIP: 888 at conference.freeswitch.org
PSTN: +1-919-386-9900

VUC IRC channel: #vuc

Come join us!
-MC
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From mike at jerris.com  Fri Mar 26 11:56:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Mar 2010 14:56:31 -0400
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap committors.
	ACTION REQUIRED
Message-ID: <E90742EF-093C-498F-9420-7B9DB6E48CBA@jerris.com>

We are in process of our migration from svn to git for our main source control.  We will likely still have a read only svn mirror, but all future commits will go to git after the migration.  In order to proceed with this process, I will need your full name, public key, email address, and svn username.  If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo.

For information on how to use git: http://git-scm.com/documentation , http://git-scm.com/course/svn.html
For information on how to generate a public key if you do not already have one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html

For general users, more information will follow on these changes.  For now, nothing has changed.  Around April 1, there will likely be a 1 time need to do a full checkout if you plan on continuing to use the svn mirror, or a fresh git checkout if you plan on switching to git.

If you have any other questions about the process or changes, feel free to ask.

Mike
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From mike at jerris.com  Fri Mar 26 12:05:37 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Mar 2010 15:05:37 -0400
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
	committors.ACTION REQUIRED
In-Reply-To: <5008F84104E54F11AC95D38CD0BD3F32@MOBILEE1705>
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
	<5008F84104E54F11AC95D38CD0BD3F32@MOBILEE1705>
Message-ID: <B2C533B4-C619-461E-8955-C2DA5370C3F5@jerris.com>

This will be transparent, with the possible exception of a one time need to do a single fresh checkout.  I am working to see if we can maintain the svn repo as based on the current repo, but this may not be possible.

On Mar 26, 2010, at 2:11 PM, Madovsky wrote:

> for update with make current it will be transparent ?
> 
> ----- Original Message ----- 
> From: "Michael Jerris" <mike at jerris.com>
> To: <freeswitch-dev at lists.freeswitch.org>; "freeswitch-users" 
> <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, March 26, 2010 1:59 PM
> Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap 
> committors.ACTION REQUIRED
> 
> 
>> We are in process of our migration from svn to git for our main source 
>> control.  We will likely still have a read only svn mirror, but all future 
>> commits will go to git after the migration.  In order to proceed with this 
>> process, I will need your full name, public key, email address, and svn 
>> username.  If you have outstanding svn branches that you would like to 
>> keep, please contact me with these details as well, I do not currently 
>> plan on merging any branches from svn to the new git repo.
>> 
>> For information on how to use git: http://git-scm.com/documentation , 
>> http://git-scm.com/course/svn.html
>> For information on how to generate a public key if you do not already have 
>> one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html
>> 
>> For general users, more information will follow on these changes.  For 
>> now, nothing has changed.  Around April 1, there will likely be a 1 time 
>> need to do a full checkout if you plan on continuing to use the svn 
>> mirror, or a fresh git checkout if you plan on switching to git.
>> 
>> If you have any other questions about the process or changes, feel free to 
>> ask.
>> 
>> Mike



From stevendt at primrosebank.net  Fri Mar 26 12:10:23 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 26 Mar 2010 19:10:23 -0000
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
	committors.ACTION REQUIRED
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
Message-ID: <3EC6DEDFF86D43BF8FDAB5B144726F7B@bp1.ad.bp.com>

Mike,

could you please add a few more words of explanation for the uninitiated 
(like ME!) . . . .

I don't contribute code (if only I could . . .), I'm on Windows, and only 
use Tortoise SVN to be able to build the current source.

>From what you say below, it sounds like the SVN stuff will still be there, 
so maybe I don't  need to change anything, but a few questions please . . .

Will the "SVN mirror" get updated as soon as the git code changes ? i.e., 
how current will it be ?
Will SVN be around for a good while yet, or is it on its way out ? Do I need 
to think about changing things now ?

What does git need for Windows users? I just had a quick look at your links 
and it looks like it needs Cygwin ?
Does that mean that I would need to learn something new (Cygwin) to be able 
to migrate to git if required ? Or is everything seamless ? - OK, maybe not 
seamless, but relatively easy ?

regards
Dave




----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-dev at lists.freeswitch.org>; "freeswitch-users" 
<freeswitch-users at lists.freeswitch.org>
Sent: Friday, March 26, 2010 5:59 PM
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap 
committors.ACTION REQUIRED


> We are in process of our migration from svn to git for our main source 
> control.  We will likely still have a read only svn mirror, but all future 
> commits will go to git after the migration.  In order to proceed with this 
> process, I will need your full name, public key, email address, and svn 
> username.  If you have outstanding svn branches that you would like to 
> keep, please contact me with these details as well, I do not currently 
> plan on merging any branches from svn to the new git repo.
>
> For information on how to use git: http://git-scm.com/documentation , 
> http://git-scm.com/course/svn.html
> For information on how to generate a public key if you do not already have 
> one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html
>
> For general users, more information will follow on these changes.  For 
> now, nothing has changed.  Around April 1, there will likely be a 1 time 
> need to do a full checkout if you plan on continuing to use the svn 
> mirror, or a fresh git checkout if you plan on switching to git.
>
> If you have any other questions about the process or changes, feel free to 
> ask.
>
> Mike
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From toofics at gmail.com  Fri Mar 26 12:40:32 2010
From: toofics at gmail.com (Victor Toofic)
Date: Fri, 26 Mar 2010 13:40:32 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
Message-ID: <1269632432.5486.54.camel@ktulu>

Hi!

Does anybody know if there is a way to "bridge" an answered channel to a
channel that is in "early media" state?

The scenario is like this:

 Channel A               Channel B
-------------------> FS ------------->
(pre_answer, park)       (answer)
<---<---<---<---<--- Audio <---<---<---

When the channel A arrives I pre_answer and park it, so it is in "early
media" state. After that I originate the channel B. When the channel B
answers I just want to relay the audio coming from B to A, I don't care
if A can't talk to B at that moment.

Is there a way to achieve this? Some ideas?

Thnks!

-  
Regards,
V?ctor Toofic




From jcasale at activenetwerx.com  Fri Mar 26 12:53:00 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 26 Mar 2010 19:53:00 +0000
Subject: [Freeswitch-users] Installing DAHDI compiled with Oslec from RPMs
	in wiki
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EBF7F@prato.activenetwerx.local>

Hey guys,

I just added my notes to the wiki:
http://wiki.freeswitch.org/wiki/Installing_DAHDI_compiled_with_Oslec_from_RPMs

Maybe some of the more seasoned guys can have a quick look for the sake of accuracy.
I have had this running for a couple days with good results.

jlc


From nandy1925 at gmail.com  Fri Mar 26 13:19:17 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sat, 27 Mar 2010 04:19:17 +0800
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
Message-ID: <7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>

hi itamar,

tks. are u happy w/ portech?

-nandy

On Fri, Mar 26, 2010 at 5:01 PM, Itamar Reis Peixoto <
itamar at ispbrasil.com.br> wrote:

> On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon <gcd at i.ph> wrote:
> > hello everybody,
> >
> > i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> > brand/model you had and your go/no-go recommendation.
> >
> > tks a lot,
> > nandy
> >
>
> portech ?
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Fri Mar 26 13:21:47 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sat, 27 Mar 2010 04:21:47 +0800
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <4BAC867B.9010503@gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<4BAC867B.9010503@gmail.com>
Message-ID: <7d0bfd8c1003261321j773eb24ax88f414adfb8baf05@mail.gmail.com>

hi nazim,

this is for future major projects. tks for the tip.

On Fri, Mar 26, 2010 at 6:03 PM, Nazim Agabekov <nazim.agabekov at gmail.com>wrote:

>  Hello Nandy
>
> I'm using Sangoma's SS7 gateway for interconnecting E1/ISUP and FreeSWITCH.
> It's rock solid and easy maintain.
> Also, Sangoma's support team is  _Very_ friendly.
>
>
> http://sangoma.com/products/software_building_blocks/ss7_solutions/ss7_signaling_and_media_gateway.html
>
>
> On 03/26/2010 09:49 AM, Nandy Dagondon wrote:
>
> hello everybody,
>
> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> brand/model you had and your go/no-go recommendation.
>
> tks a lot,
> nandy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nandy1925 at gmail.com  Fri Mar 26 13:30:04 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sat, 27 Mar 2010 04:30:04 +0800
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <4BACF9B5.70803@puzzled.xs4all.nl>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
	<4BACF9B5.70803@puzzled.xs4all.nl>
Message-ID: <7d0bfd8c1003261330i4dcdef22q2f2df2b027db9de2@mail.gmail.com>

hi patrick,  tks.  it's the first brand i know.

On Sat, Mar 27, 2010 at 2:15 AM, Patrick
<freeswitch-list at puzzled.xs4all.nl>wrote:

> On 03/26/2010 10:01 AM, Itamar Reis Peixoto wrote:
> > On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon<gcd at i.ph>  wrote:
> >> hello everybody,
> >>
> >> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> >> brand/model you had and your go/no-go recommendation.
> >>
> >> tks a lot,
> >> nandy
> >>
> >
> > portech ?
>
> 2N VoiceBlue. I only know they work fine with Asterisk. Iirc they are
> more expensive than the Portech boxes.
>
> See http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway.html
>
> Regards,
> Patrick
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Mar 26 14:18:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 26 Mar 2010 17:18:40 -0400
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
	committors.ACTION REQUIRED
In-Reply-To: <3EC6DEDFF86D43BF8FDAB5B144726F7B@bp1.ad.bp.com>
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
	<3EC6DEDFF86D43BF8FDAB5B144726F7B@bp1.ad.bp.com>
Message-ID: <A5204C08-94A5-468D-9DAF-8C8498ECE891@jerris.com>



On Mar 26, 2010, at 3:10 PM, "Dave Stevenson"  
<stevendt at primrosebank.net> wrote:

> Mike,
>
> could you please add a few more words of explanation for the  
> uninitiated
> (like ME!) . . . .
>
> I don't contribute code (if only I could . . .), I'm on Windows, and  
> only
> use Tortoise SVN to be able to build the current source.
>
>> From what you say below, it sounds like the SVN stuff will still be  
>> there,
> so maybe I don't  need to change anything, but a few questions  
> please . . .
>
> Will the "SVN mirror" get updated as soon as the git code changes ?  
> i.e.,
> how current will it be ?

it will be updated immidiately, however in the future we are likely to  
make new branches for work on unstable new features and these features  
would require git to access until they are merged into the main tree.   
this will allow us to maintain our relative stability of our trunk  
while still allowing collaboritive work on new features.


> Will SVN be around for a good while yet, or is it on its way out ?  
> Do I need
> to think about changing things now ?
>

at this time we plan to maintain the mirror of the master branch (svn  
trunk) for quite some time or forever.  branches and tags would only  
be maintained in git but would continue to be available as a tarball.

> What does git need for Windows users? I just had a quick look at  
> your links
> and it looks like it needs Cygwin ?
> Does that mean that I would need to learn something new (Cygwin) to  
> be able
> to migrate to git if required ? Or is everything seamless ? - OK,  
> maybe not
> seamless, but relatively easy ?

there is a git client that does not require cygwin called tortisegit,  
that is very much like the svn version.  I have no intention of ever  
adding any dependency on cygwin.

>
> regards
> Dave
>
>
>
>
> ----- Original Message -----
> From: "Michael Jerris" <mike at jerris.com>
> To: <freeswitch-dev at lists.freeswitch.org>; "freeswitch-users"
> <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, March 26, 2010 5:59 PM
> Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
> committors.ACTION REQUIRED
>
>
>> We are in process of our migration from svn to git for our main  
>> source
>> control.  We will likely still have a read only svn mirror, but all  
>> future
>> commits will go to git after the migration.  In order to proceed  
>> with this
>> process, I will need your full name, public key, email address, and  
>> svn
>> username.  If you have outstanding svn branches that you would like  
>> to
>> keep, please contact me with these details as well, I do not  
>> currently
>> plan on merging any branches from svn to the new git repo.
>>
>> For information on how to use git: http://git-scm.com/documentation ,
>> http://git-scm.com/course/svn.html
>> For information on how to generate a public key if you do not  
>> already have
>> one:  http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html
>>
>> For general users, more information will follow on these changes.   
>> For
>> now, nothing has changed.  Around April 1, there will likely be a 1  
>> time
>> need to do a full checkout if you plan on continuing to use the svn
>> mirror, or a fresh git checkout if you plan on switching to git.
>>
>> If you have any other questions about the process or changes, feel  
>> free to
>> ask.
>>
>> Mike
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From edpimentl at gmail.com  Fri Mar 26 14:35:03 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 26 Mar 2010 17:35:03 -0400
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com> 
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com> 
	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
Message-ID: <9dc4a1671003261435h11713a5v3aa03e05652a46f8@mail.gmail.com>

I have had no issues with portech.
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From jcasale at activenetwerx.com  Fri Mar 26 14:36:08 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 26 Mar 2010 21:36:08 +0000
Subject: [Freeswitch-users] Call Park issue
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>

I just tried the call park based on the recent thread using the two blocks
from http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Valet_Autoselect_Parking_Stall
and sometimes it works as expected, and sometimes the caller that parks the call gets the
MOH when they try to unpark the call when dialing the ext of the call they
parked?

Any idea what I am doing wrong?

Thanks,
jlc


From itamar at ispbrasil.com.br  Fri Mar 26 14:41:43 2010
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 26 Mar 2010 18:41:43 -0300
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
Message-ID: <d3ceea381003261441j12f86a05ld850f09aee1e6870@mail.gmail.com>

On Fri, Mar 26, 2010 at 5:19 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> hi itamar,
>
> tks. are u happy w/ portech?
>
> -nandy

yes, very good.



------------

Itamar Reis Peixoto

e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599


From anthony.minessale at gmail.com  Fri Mar 26 14:57:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 16:57:11 -0500
Subject: [Freeswitch-users] Call Park issue
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
Message-ID: <191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>

is the caller hungup? if you call the ext when it's free you will
essentially be parking yourself.


On Fri, Mar 26, 2010 at 4:36 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> I just tried the call park based on the recent thread using the two blocks
> from
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park#Valet_Autoselect_Parking_Stall
> and sometimes it works as expected, and sometimes the caller that parks the
> call gets the
> MOH when they try to unpark the call when dialing the ext of the call they
> parked?
>
> Any idea what I am doing wrong?
>
> Thanks,
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From 12ukwn at gmail.com  Fri Mar 26 14:57:18 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Fri, 26 Mar 2010 22:57:18 +0100
Subject: [Freeswitch-users] suggestions
Message-ID: <20100326225718.7902f354@anubis.defcon1>

Hi,

what is the best way to make requests for features &| suggestions?

-- 
Given my druthers, I'd druther not.


From jcasale at activenetwerx.com  Fri Mar 26 15:10:42 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 26 Mar 2010 22:10:42 +0000
Subject: [Freeswitch-users] Call Park issue
In-Reply-To: <191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
	<191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EC3BE@prato.activenetwerx.local>

>is the caller hungup? if you call the ext when it's free you will essentially be parking yourself.

Nope, it was me standing next to the person with my cell. I
called in, they answered, they parked me (we both heard "8501"), they
hung up while I remained on listening to MOH, but when they dialed
8501 my MOH stopped, and they got it!

Weird...

Thanks,
jlc



From toofics at gmail.com  Fri Mar 26 15:23:32 2010
From: toofics at gmail.com (Victor Toofic)
Date: Fri, 26 Mar 2010 16:23:32 -0600
Subject: [Freeswitch-users] One Way Bridge Channels
Message-ID: <1269642212.5486.62.camel@ktulu>

Hi!

Does anybody know if there is a way to "bridge" an answered channel to a
channel that is in "early media" state?

The scenario is like this:

 Channel A               Channel B
-------------------> FS ------------->
(pre_answer, park)       (answer)
<---<---<---<---<--- Audio <---<---<---

When the channel A arrives I pre_answer and park it, so it is in "early
media" state. After that I originate the channel B. When the channel B
answers I just want to relay the audio coming from B to A, I don't care
if A can't talk to B at that moment.

Is there a way to achieve this? Some ideas?

Thnks!

-  
Regards,
V?ctor Toofic



From anthony.minessale at gmail.com  Fri Mar 26 15:28:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 17:28:46 -0500
Subject: [Freeswitch-users] Call Park issue
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E98EC3BE@prato.activenetwerx.local>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
	<191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98EC3BE@prato.activenetwerx.local>
Message-ID: <191c3a031003261528t31c704x88a9e45504564b93@mail.gmail.com>

you should reproduce that with the console logging up

enter this command at the CLI:

console loglevel debug

then reproduce and post the results



On Fri, Mar 26, 2010 at 5:10 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >is the caller hungup? if you call the ext when it's free you will
> essentially be parking yourself.
>
> Nope, it was me standing next to the person with my cell. I
> called in, they answered, they parked me (we both heard "8501"), they
> hung up while I remained on listening to MOH, but when they dialed
> 8501 my MOH stopped, and they got it!
>
> Weird...
>
> Thanks,
> jlc
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 26 15:29:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 17:29:57 -0500
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100326225718.7902f354@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
Message-ID: <191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>

you can post ideas here for discussion.
if you want something specific you can open a jira incident at
http://jira.freeswitch.org under "improvement" category.



On Fri, Mar 26, 2010 at 4:57 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Hi,
>
> what is the best way to make requests for features &| suggestions?
>
> --
> Given my druthers, I'd druther not.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 26 15:31:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 17:31:51 -0500
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <1269632432.5486.54.camel@ktulu>
References: <1269632432.5486.54.camel@ktulu>
Message-ID: <191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>

yes,

don't park it, have it dial the other leg with the bridge app.

originate <dialstring1> bridge:<dialstring2> inline


On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic <toofics at gmail.com> wrote:

> Hi!
>
> Does anybody know if there is a way to "bridge" an answered channel to a
> channel that is in "early media" state?
>
> The scenario is like this:
>
>  Channel A               Channel B
> -------------------> FS ------------->
> (pre_answer, park)       (answer)
> <---<---<---<---<--- Audio <---<---<---
>
> When the channel A arrives I pre_answer and park it, so it is in "early
> media" state. After that I originate the channel B. When the channel B
> answers I just want to relay the audio coming from B to A, I don't care
> if A can't talk to B at that moment.
>
> Is there a way to achieve this? Some ideas?
>
> Thnks!
>
> -
> Regards,
> V?ctor Toofic
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jcasale at activenetwerx.com  Fri Mar 26 15:38:53 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 26 Mar 2010 22:38:53 +0000
Subject: [Freeswitch-users] Call Park issue
In-Reply-To: <191c3a031003261528t31c704x88a9e45504564b93@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
	<191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98EC3BE@prato.activenetwerx.local>
	<191c3a031003261528t31c704x88a9e45504564b93@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EC459@prato.activenetwerx.local>

>you should reproduce that with the console logging up
>
>enter this command at the CLI:
>
>console loglevel debug 
>
>then reproduce and post the results 

Sure thing, I'll drive back in on the weekend.
jlc


From anthony.minessale at gmail.com  Fri Mar 26 15:48:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 17:48:27 -0500
Subject: [Freeswitch-users] Bad Video in latest SVN
In-Reply-To: <201003261323.50280.yivzhenko@mksat.net>
References: <201003261323.50280.yivzhenko@mksat.net>
Message-ID: <191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>

what exactly are you doing?
can you provide the dialplan and log of the call you are having problems
with.
the more information the better


On Fri, Mar 26, 2010 at 6:23 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:

> Hi all,
>
> After upgrade from SVN 16013 to latest SVN 17109
> Video quality become very bad.
> I'm use H264 video codec and Grandstream GXV3140 phones
>
> Anyone knows why this may happen?
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From jason at jasonjgw.net  Fri Mar 26 16:03:31 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 27 Mar 2010 10:03:31 +1100
Subject: [Freeswitch-users] Attention all FreeSWITCH and Openzap
 committors. ACTION REQUIRED
In-Reply-To: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
References: <E4E298E0-1A16-4FEA-B592-A337DAFD7707@Jerris.com>
Message-ID: <20100326230330.GA5485@jdc.jasonjgw.net>

Michael Jerris <mike at jerris.com> wrote:
> We are in process of our migration from svn to git for our main source control.  

Excellent. for those who are new to Git, there is a very informative and well
written book available here and in hard copy from your favourite online book
seller:
http://progit.org/book/

I am sure the Git URL for the repository will be announced here when it is
ready, at which point we can clone it and proceed from there.



From 12ukwn at gmail.com  Fri Mar 26 16:12:50 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 00:12:50 +0100
Subject: [Freeswitch-users] suggestions
In-Reply-To: <191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>
References: <20100326225718.7902f354@anubis.defcon1>
	<191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>
Message-ID: <20100327001250.3f72bf73@anubis.defcon1>

Le Fri, 26 Mar 2010 17:29:57 -0500,
Anthony Minessale <anthony.minessale at gmail.com> a ?crit :

Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is to 
have some more features:

* Connection fee

* Time-credit (from french term "cr?dit-temps", means when you make a
  call you're first invoiced a minimum of X seconds - sometimes along a
  connection fee:(

* A per second invoicing instead of per minute, but without a DB call every
  second, just one call at hangup.


> you can post ideas here for discussion.
> if you want something specific you can open a jira incident at
> http://jira.freeswitch.org under "improvement" category.

-- 
Conscious is when you are aware of something and conscience is when you
wish you weren't.


From sos at sokhapkin.dyndns.org  Fri Mar 26 16:38:01 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 26 Mar 2010 19:38:01 -0400
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100327001250.3f72bf73@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
	<191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>
	<20100327001250.3f72bf73@anubis.defcon1>
Message-ID: <201003261938.01670.sos@sokhapkin.dyndns.org>

In a2billing terms items 2 and 3 are "init block" and "billing block", i.e. 
minimum call duration to bill and billing increment after "init block" seconds 
(in your case you want billing block to be 1 second). Sure these essential 
billing features are missed in  mod_nibblebill. Here are 4 "must have" billing 
parameters for  mod_nibblebill:

1. Connection fee
2. sell rate
3. init block
4. billing increment

If call duration is less than "init block" seconds, then call should be 
charged for "init block" seconds, otherwise for actual call length taking into 
account "billing increment" value.

On Friday 26 March 2010, Jean-Yves F. Barbier wrote:
> Le Fri, 26 Mar 2010 17:29:57 -0500,
> Anthony Minessale <anthony.minessale at gmail.com> a ?crit :
> 
> Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is to
> have some more features:
> 
> * Connection fee
> 
> * Time-credit (from french term "cr?dit-temps", means when you make a
>   call you're first invoiced a minimum of X seconds - sometimes along a
>   connection fee:(
> 
> * A per second invoicing instead of per minute, but without a DB call every
>   second, just one call at hangup.
> 
> > you can post ideas here for discussion.
> > if you want something specific you can open a jira incident at
> > http://jira.freeswitch.org under "improvement" category.
> 



From sos at sokhapkin.dyndns.org  Fri Mar 26 16:55:11 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 26 Mar 2010 19:55:11 -0400
Subject: [Freeswitch-users] suggestions
In-Reply-To: <201003261938.01670.sos@sokhapkin.dyndns.org>
References: <20100326225718.7902f354@anubis.defcon1>
	<20100327001250.3f72bf73@anubis.defcon1>
	<201003261938.01670.sos@sokhapkin.dyndns.org>
Message-ID: <201003261955.11318.sos@sokhapkin.dyndns.org>

Here is lua code I use instead of mod_nibblebill:

                if(ANSWEREDTIME<=initblock) then
                        CHARGETIME=initblock
                else
                        incrs=math.floor((ANSWEREDTIME-
initblock+billingblock-1)/billingblock);
                        CHARGETIME=initblock+incrs*billingblock;
                end
                sessionbill=(CHARGETIME*calledrate)/60 + connfee;


On Friday 26 March 2010, Sergey Okhapkin wrote:
> In a2billing terms items 2 and 3 are "init block" and "billing block", i.e.
> minimum call duration to bill and billing increment after "init block"
>  seconds (in your case you want billing block to be 1 second). Sure these
>  essential billing features are missed in  mod_nibblebill. Here are 4 "must
>  have" billing parameters for  mod_nibblebill:
> 
> 1. Connection fee
> 2. sell rate
> 3. init block
> 4. billing increment
> 
> If call duration is less than "init block" seconds, then call should be
> charged for "init block" seconds, otherwise for actual call length taking
>  into account "billing increment" value.
> 
> On Friday 26 March 2010, Jean-Yves F. Barbier wrote:
> > Le Fri, 26 Mar 2010 17:29:57 -0500,
> > Anthony Minessale <anthony.minessale at gmail.com> a ?crit :
> >
> > Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is to
> > have some more features:
> >
> > * Connection fee
> >
> > * Time-credit (from french term "cr?dit-temps", means when you make a
> >   call you're first invoiced a minimum of X seconds - sometimes along a
> >   connection fee:(
> >
> > * A per second invoicing instead of per minute, but without a DB call
> > every second, just one call at hangup.
> >
> > > you can post ideas here for discussion.
> > > if you want something specific you can open a jira incident at
> > > http://jira.freeswitch.org under "improvement" category.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From rupa at rupa.com  Fri Mar 26 17:02:19 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 26 Mar 2010 19:02:19 -0500
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100327001250.3f72bf73@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
	<191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>
	<20100327001250.3f72bf73@anubis.defcon1>
Message-ID: <b39273531003261702k35bb788fycc1595aa237e578b@mail.gmail.com>

sell_rate you can get as a user_rate from mod_lcr or do your
own separate query if you don't want lcr info too.

billing increment, I've seen a patch that implements this.  I was gonna port
that forward to current rev.  If Darren or someone else doesn't get it first
I'll get to it eventually.

init_block - would implement at same time as billing_increment.

connection_fee - again, like init_block.

In fact, you *could* get all the values you need via the version of mod_lcr
that is sitting locally since you can query/set arbitrary channel vars via
custom sql.  You can decide how these fields relate to your lcr tables (they
may not, it is up to you based on the sql you craft).

You also mentioned you don't want to bill in real-time but only at end of
call.  This you can do today by turning off the heartbeat.   I think this is
already discussed on the wiki.

As always, a bounty goes a long way.   One of the devs might jump on it
sooner if there is money in it.  I'm swamped for the next week at least
so...  won't  have a chance to even look at this until then.  Not to mention
I want to finish cleaning up my limit stuff.

On Fri, Mar 26, 2010 at 6:12 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Le Fri, 26 Mar 2010 17:29:57 -0500,
> Anthony Minessale <anthony.minessale at gmail.com> a ?crit :
>
> Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is to
> have some more features:
>
> * Connection fee
>
> * Time-credit (from french term "cr?dit-temps", means when you make a
>  call you're first invoiced a minimum of X seconds - sometimes along a
>  connection fee:(
>
> * A per second invoicing instead of per minute, but without a DB call every
>  second, just one call at hangup.
>
>
> > you can post ideas here for discussion.
> > if you want something specific you can open a jira incident at
> > http://jira.freeswitch.org under "improvement" category.
>
> --
> Conscious is when you are aware of something and conscience is when you
> wish you weren't.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From 12ukwn at gmail.com  Fri Mar 26 17:11:33 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 01:11:33 +0100
Subject: [Freeswitch-users] suggestions
In-Reply-To: <201003261955.11318.sos@sokhapkin.dyndns.org>
References: <20100326225718.7902f354@anubis.defcon1>
	<20100327001250.3f72bf73@anubis.defcon1>
	<201003261938.01670.sos@sokhapkin.dyndns.org>
	<201003261955.11318.sos@sokhapkin.dyndns.org>
Message-ID: <20100327011133.26dd8262@anubis.defcon1>

Le Fri, 26 Mar 2010 19:55:11 -0400,
Sergey Okhapkin <sos at sokhapkin.dyndns.org> a ?crit :

> Here is lua code I use instead of mod_nibblebill:
> 
>                 if(ANSWEREDTIME<=initblock) then
>                         CHARGETIME=initblock
>                 else
>                         incrs=math.floor((ANSWEREDTIME-
> initblock+billingblock-1)/billingblock);
>                         CHARGETIME=initblock+incrs*billingblock;
>                 end
>                 sessionbill=(CHARGETIME*calledrate)/60 + connfee;

I realise 2 things:
1- FS is even more powerful than all I read about it,
2- It's gonna take me "some" time to fully understand it...

-- 
Dogs just don't seem to be able to tell the difference between important
people and the rest of us.


From toofics at gmail.com  Fri Mar 26 17:12:14 2010
From: toofics at gmail.com (Victor Toofic)
Date: Fri, 26 Mar 2010 18:12:14 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
Message-ID: <1269648734.5486.93.camel@ktulu>

I don't get it. What is "<dialstring2>" for? I'm not originating both
legs of the call.

I already have the incoming A leg in the switch and I want it to stay in
early media and "bridge" it one-way only to a originated B leg that is
always answered since the very first second.

What I am trying to do is some kind of call progress supervision on the
B leg and answer and bridge to the A leg only when I detect the B leg
was really answered. Before that I just want to pass the audio coming
from the answered B leg to the early-media-state A leg.. so the A leg
can hear the ringback tones, voice-mail messages, etc.

I already detect the tones and when the B leg is really answered, at
that moment I can do an uuid_bridge of both uuids and get the legs
bridged. But I don't see how I could pass audio from B to A before
answering the A leg.

Is this possible in any way with FS??

Maybe I didn't understand you, but I don't see how could it be possible
to do it with an inline dialplan.

Thnks!!


On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
> yes, 
> 
> don't park it, have it dial the other leg with the bridge app.
> 
> originate <dialstring1> bridge:<dialstring2> inline
> 
> 
> On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic <toofics at gmail.com>
> wrote:
>         Hi!
>         
>         Does anybody know if there is a way to "bridge" an answered
>         channel to a
>         channel that is in "early media" state?
>         
>         The scenario is like this:
>         
>          Channel A               Channel B
>         -------------------> FS ------------->
>         (pre_answer, park)       (answer)
>         <---<---<---<---<--- Audio <---<---<---
>         
>         When the channel A arrives I pre_answer and park it, so it is
>         in "early
>         media" state. After that I originate the channel B. When the
>         channel B
>         answers I just want to relay the audio coming from B to A, I
>         don't care
>         if A can't talk to B at that moment.
>         
>         Is there a way to achieve this? Some ideas?
>         
>         Thnks!
>         
>         -
>         Regards,
>         V?ctor Toofic
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sos at sokhapkin.dyndns.org  Fri Mar 26 17:13:01 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 26 Mar 2010 20:13:01 -0400
Subject: [Freeswitch-users] suggestions
In-Reply-To: <b39273531003261702k35bb788fycc1595aa237e578b@mail.gmail.com>
References: <20100326225718.7902f354@anubis.defcon1>
	<20100327001250.3f72bf73@anubis.defcon1>
	<b39273531003261702k35bb788fycc1595aa237e578b@mail.gmail.com>
Message-ID: <201003262013.01724.sos@sokhapkin.dyndns.org>

Could you give more info about "since you can query/set arbitrary channel vars 
via custom sql"? How can I set a channel variable using mod_lcr with custom 
sql query?

On Friday 26 March 2010, Rupa Schomaker wrote:
> sell_rate you can get as a user_rate from mod_lcr or do your
> own separate query if you don't want lcr info too.
> 
> billing increment, I've seen a patch that implements this.  I was gonna
>  port that forward to current rev.  If Darren or someone else doesn't get
>  it first I'll get to it eventually.
> 
> init_block - would implement at same time as billing_increment.
> 
> connection_fee - again, like init_block.
> 
> In fact, you *could* get all the values you need via the version of mod_lcr
> that is sitting locally since you can query/set arbitrary channel vars via
> custom sql.  You can decide how these fields relate to your lcr tables
>  (they may not, it is up to you based on the sql you craft).
> 
> You also mentioned you don't want to bill in real-time but only at end of
> call.  This you can do today by turning off the heartbeat.   I think this
>  is already discussed on the wiki.
> 
> As always, a bounty goes a long way.   One of the devs might jump on it
> sooner if there is money in it.  I'm swamped for the next week at least
> so...  won't  have a chance to even look at this until then.  Not to
>  mention I want to finish cleaning up my limit stuff.
> 
> On Fri, Mar 26, 2010 at 6:12 PM, Jean-Yves F. Barbier 
<12ukwn at gmail.com>wrote:
> > Le Fri, 26 Mar 2010 17:29:57 -0500,
> > Anthony Minessale <anthony.minessale at gmail.com> a ?crit :
> >
> > Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is to
> > have some more features:
> >
> > * Connection fee
> >
> > * Time-credit (from french term "cr?dit-temps", means when you make a
> >  call you're first invoiced a minimum of X seconds - sometimes along a
> >  connection fee:(
> >
> > * A per second invoicing instead of per minute, but without a DB call
> > every second, just one call at hangup.
> >
> > > you can post ideas here for discussion.
> > > if you want something specific you can open a jira incident at
> > > http://jira.freeswitch.org under "improvement" category.
> >
> > --
> > Conscious is when you are aware of something and conscience is when you
> > wish you weren't.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From msc at freeswitch.org  Fri Mar 26 17:16:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Mar 2010 17:16:59 -0700
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100327011133.26dd8262@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
	<20100327001250.3f72bf73@anubis.defcon1>
	<201003261938.01670.sos@sokhapkin.dyndns.org>
	<201003261955.11318.sos@sokhapkin.dyndns.org>
	<20100327011133.26dd8262@anubis.defcon1>
Message-ID: <87f2f3b91003261716v4c2828e1k9959c94a86247a97@mail.gmail.com>

On Fri, Mar 26, 2010 at 5:11 PM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Le Fri, 26 Mar 2010 19:55:11 -0400,
> Sergey Okhapkin <sos at sokhapkin.dyndns.org> a ?crit :
>
> > Here is lua code I use instead of mod_nibblebill:
> >
> >                 if(ANSWEREDTIME<=initblock) then
> >                         CHARGETIME=initblock
> >                 else
> >                         incrs=math.floor((ANSWEREDTIME-
> > initblock+billingblock-1)/billingblock);
> >                         CHARGETIME=initblock+incrs*billingblock;
> >                 end
> >                 sessionbill=(CHARGETIME*calledrate)/60 + connfee;
>
> I realise 2 things:
> 1- FS is even more powerful than all I read about it,
>
True

> 2- It's gonna take me "some" time to fully understand it...
>
Also true.

It's totally worth the investment in time and energy. Enjoy the ride!
-MC
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From sos at sokhapkin.dyndns.org  Fri Mar 26 17:28:44 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 26 Mar 2010 20:28:44 -0400
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100327011133.26dd8262@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
	<201003261955.11318.sos@sokhapkin.dyndns.org>
	<20100327011133.26dd8262@anubis.defcon1>
Message-ID: <201003262028.44360.sos@sokhapkin.dyndns.org>

I used asterisk for years (and still use it for voice mail and access numbers 
and call back), but switched regular outgoing calls to FS. And I don't want to 
look back at asterisk. * doesn't have the flexibility FS offers and I need.

On Friday 26 March 2010, Jean-Yves F. Barbier wrote:
> Le Fri, 26 Mar 2010 19:55:11 -0400,
> 
> Sergey Okhapkin <sos at sokhapkin.dyndns.org> a ?crit :
> > Here is lua code I use instead of mod_nibblebill:
> >
> >                 if(ANSWEREDTIME<=initblock) then
> >                         CHARGETIME=initblock
> >                 else
> >                         incrs=math.floor((ANSWEREDTIME-
> > initblock+billingblock-1)/billingblock);
> >                         CHARGETIME=initblock+incrs*billingblock;
> >                 end
> >                 sessionbill=(CHARGETIME*calledrate)/60 + connfee;
> 
> I realise 2 things:
> 1- FS is even more powerful than all I read about it,
> 2- It's gonna take me "some" time to fully understand it...
> 



From anthony.minessale at gmail.com  Fri Mar 26 17:31:00 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 19:31:00 -0500
Subject: [Freeswitch-users] Bad Video in latest SVN
In-Reply-To: <191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>
References: <201003261323.50280.yivzhenko@mksat.net>
	<191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>
Message-ID: <191c3a031003261731u25b6ca11v3421f36111b9fc64@mail.gmail.com>

Never mind, I'm too impatient.
Can you update to latest trunk 17117 and see if it's better?


On Fri, Mar 26, 2010 at 5:48 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> what exactly are you doing?
> can you provide the dialplan and log of the call you are having problems
> with.
> the more information the better
>
>
>
> On Fri, Mar 26, 2010 at 6:23 AM, Yuriy Ivzhenko <yivzhenko at mksat.net>wrote:
>
>> Hi all,
>>
>> After upgrade from SVN 16013 to latest SVN 17109
>> Video quality become very bad.
>> I'm use H264 video codec and Grandstream GXV3140 phones
>>
>> Anyone knows why this may happen?
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From anthony.minessale at gmail.com  Fri Mar 26 17:33:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Mar 2010 19:33:54 -0500
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <1269648734.5486.93.camel@ktulu>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
Message-ID: <191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>

no,

That sounds a little over-thought though?
Why can't you just use the bridge app?  We have extensive parameters to do
progress supervision on the wiki.


On Fri, Mar 26, 2010 at 7:12 PM, Victor Toofic <toofics at gmail.com> wrote:

> I don't get it. What is "<dialstring2>" for? I'm not originating both
> legs of the call.
>
> I already have the incoming A leg in the switch and I want it to stay in
> early media and "bridge" it one-way only to a originated B leg that is
> always answered since the very first second.
>
> What I am trying to do is some kind of call progress supervision on the
> B leg and answer and bridge to the A leg only when I detect the B leg
> was really answered. Before that I just want to pass the audio coming
> from the answered B leg to the early-media-state A leg.. so the A leg
> can hear the ringback tones, voice-mail messages, etc.
>
> I already detect the tones and when the B leg is really answered, at
> that moment I can do an uuid_bridge of both uuids and get the legs
> bridged. But I don't see how I could pass audio from B to A before
> answering the A leg.
>
> Is this possible in any way with FS??
>
> Maybe I didn't understand you, but I don't see how could it be possible
> to do it with an inline dialplan.
>
> Thnks!!
>
>
> On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
> > yes,
> >
> > don't park it, have it dial the other leg with the bridge app.
> >
> > originate <dialstring1> bridge:<dialstring2> inline
> >
> >
> > On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic <toofics at gmail.com>
> > wrote:
> >         Hi!
> >
> >         Does anybody know if there is a way to "bridge" an answered
> >         channel to a
> >         channel that is in "early media" state?
> >
> >         The scenario is like this:
> >
> >          Channel A               Channel B
> >         -------------------> FS ------------->
> >         (pre_answer, park)       (answer)
> >         <---<---<---<---<--- Audio <---<---<---
> >
> >         When the channel A arrives I pre_answer and park it, so it is
> >         in "early
> >         media" state. After that I originate the channel B. When the
> >         channel B
> >         answers I just want to relay the audio coming from B to A, I
> >         don't care
> >         if A can't talk to B at that moment.
> >
> >         Is there a way to achieve this? Some ideas?
> >
> >         Thnks!
> >
> >         -
> >         Regards,
> >         V?ctor Toofic
> >
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From 12ukwn at gmail.com  Fri Mar 26 17:49:10 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 01:49:10 +0100
Subject: [Freeswitch-users] suggestions
In-Reply-To: <201003262028.44360.sos@sokhapkin.dyndns.org>
References: <20100326225718.7902f354@anubis.defcon1>
	<201003261955.11318.sos@sokhapkin.dyndns.org>
	<20100327011133.26dd8262@anubis.defcon1>
	<201003262028.44360.sos@sokhapkin.dyndns.org>
Message-ID: <20100327014910.596f8ab1@anubis.defcon1>

Le Fri, 26 Mar 2010 20:28:44 -0400,
Sergey Okhapkin <sos at sokhapkin.dyndns.org> a ?crit :

I also used it for some years (no DB), but I don't wanna hear about it
anymore: crashes, choppy sound, not to mention changes of programming
primitives between versions...

> I used asterisk for years (and still use it for voice mail and access
> numbers and call back), but switched regular outgoing calls to FS. And I
> don't want to look back at asterisk. * doesn't have the flexibility FS
> offers and I need.

-- 
Without freedom of choice there is no creativity.
		-- Kirk, "The return of the Archons", stardate 3157.4


From toofics at gmail.com  Fri Mar 26 17:52:58 2010
From: toofics at gmail.com (Victor Toofic)
Date: Fri, 26 Mar 2010 18:52:58 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
Message-ID: <1269651178.5486.102.camel@ktulu>

Because the B leg is always immediately answered, it is giving false
answer. The ringback, busy and fastbusy tones are being played once it
is answered. And if I use the bridge app.. once the B leg gets answered
it answers the A leg.

On Fri, 2010-03-26 at 19:33 -0500, Anthony Minessale wrote:
> no,
> 
> That sounds a little over-thought though?
> Why can't you just use the bridge app?  We have extensive parameters
> to do progress supervision on the wiki.
> 
> 
> On Fri, Mar 26, 2010 at 7:12 PM, Victor Toofic <toofics at gmail.com>
> wrote:
>         I don't get it. What is "<dialstring2>" for? I'm not
>         originating both
>         legs of the call.
>         
>         I already have the incoming A leg in the switch and I want it
>         to stay in
>         early media and "bridge" it one-way only to a originated B leg
>         that is
>         always answered since the very first second.
>         
>         What I am trying to do is some kind of call progress
>         supervision on the
>         B leg and answer and bridge to the A leg only when I detect
>         the B leg
>         was really answered. Before that I just want to pass the audio
>         coming
>         from the answered B leg to the early-media-state A leg.. so
>         the A leg
>         can hear the ringback tones, voice-mail messages, etc.
>         
>         I already detect the tones and when the B leg is really
>         answered, at
>         that moment I can do an uuid_bridge of both uuids and get the
>         legs
>         bridged. But I don't see how I could pass audio from B to A
>         before
>         answering the A leg.
>         
>         Is this possible in any way with FS??
>         
>         Maybe I didn't understand you, but I don't see how could it be
>         possible
>         to do it with an inline dialplan.
>         
>         Thnks!!
>         
>         
>         
>         On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
>         > yes,
>         >
>         > don't park it, have it dial the other leg with the bridge
>         app.
>         >
>         > originate <dialstring1> bridge:<dialstring2> inline
>         >
>         >
>         > On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic
>         <toofics at gmail.com>
>         > wrote:
>         >         Hi!
>         >
>         >         Does anybody know if there is a way to "bridge" an
>         answered
>         >         channel to a
>         >         channel that is in "early media" state?
>         >
>         >         The scenario is like this:
>         >
>         >          Channel A               Channel B
>         >         -------------------> FS ------------->
>         >         (pre_answer, park)       (answer)
>         >         <---<---<---<---<--- Audio <---<---<---
>         >
>         >         When the channel A arrives I pre_answer and park it,
>         so it is
>         >         in "early
>         >         media" state. After that I originate the channel B.
>         When the
>         >         channel B
>         >         answers I just want to relay the audio coming from B
>         to A, I
>         >         don't care
>         >         if A can't talk to B at that moment.
>         >
>         >         Is there a way to achieve this? Some ideas?
>         >
>         >         Thnks!
>         >
>         >         -
>         >         Regards,
>         >         V?ctor Toofic
>         >
>         >
>         >
>         >         _______________________________________________
>         >         FreeSWITCH-users mailing list
>         >         FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org
>         >
>         >
>         >
>         > --
>         > Anthony Minessale II
>         >
>         > FreeSWITCH http://www.freeswitch.org/
>         > ClueCon http://www.cluecon.com/
>         > Twitter: http://twitter.com/FreeSWITCH_wire
>         >
>         > AIM: anthm
>         > MSN:anthony_minessale at hotmail.com
>         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         > IRC: irc.freenode.net #freeswitch
>         >
>         > FreeSWITCH Developer Conference
>         > sip:888 at conference.freeswitch.org
>         > googletalk:conf+888 at conference.freeswitch.org
>         > pstn:+19193869900
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From 12ukwn at gmail.com  Fri Mar 26 20:02:07 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 04:02:07 +0100
Subject: [Freeswitch-users] session nb
Message-ID: <20100327040207.33ac6858@anubis.defcon1>

Hi list,

On my old athlonXP-2600+, FS sessions max is 1000 & session limit is 30
(I guess connection per second?.)

How does it translate in the real world? 
Is it going down a lot if I've got many transcoding channels (let's say
iLBC internal transcoding to G711 external)
I know it couldn't be a sharp answer, but I need to have an idea of what is
possible (or not.)

-- 
This fortune is false.


From neilp at cs.stanford.edu  Fri Mar 26 20:16:31 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Fri, 26 Mar 2010 20:16:31 -0700
Subject: [Freeswitch-users] luasql on CentOS 5.4
Message-ID: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>

I can't get luasql to compile on CentOS 5.4:

[root at topaz luasql-2.1.1]# make
gcc -fPIC -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi
-pedantic -I../compat/src -I/usr/include/mysql -I/usr/local/include  -o
src/mysql.so -shared  src/luasql.o src/ls_mysql.o  -L/usr/local/mysql/lib
-lmysqlclient -lz
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol'
can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/mysql.so] Error 1

I've tried adding the -fPIC in the make but still no luck. Also, the FS wiki
warns agains compiling with the MySQL driver since it leaks memory. So how
do I install luasql for MySQL? Couldn't find any RPMs, can anyone here share
one?

Thanks,
Neil
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From rupa at rupa.com  Fri Mar 26 20:28:39 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 26 Mar 2010 22:28:39 -0500
Subject: [Freeswitch-users] suggestions
In-Reply-To: <201003262013.01724.sos@sokhapkin.dyndns.org>
References: <20100326225718.7902f354@anubis.defcon1>
	<20100327001250.3f72bf73@anubis.defcon1>
	<b39273531003261702k35bb788fycc1595aa237e578b@mail.gmail.com>
	<201003262013.01724.sos@sokhapkin.dyndns.org>
Message-ID: <b39273531003262028sa4737c0t7f6c2b1198da3094@mail.gmail.com>

You can see an (older) version of my work-in-progress along with some
discussion of how it works at:

http://jira.freeswitch.org/browse/MODAPP-340

<http://jira.freeswitch.org/browse/MODAPP-340>I haven't updated the patch
since the version in my tree also depends on my limit work.  But the
discussion should give you a better idea of how this will work.

On Fri, Mar 26, 2010 at 7:13 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Could you give more info about "since you can query/set arbitrary channel
> vars
> via custom sql"? How can I set a channel variable using mod_lcr with custom
> sql query?
>
> On Friday 26 March 2010, Rupa Schomaker wrote:
> > sell_rate you can get as a user_rate from mod_lcr or do your
> > own separate query if you don't want lcr info too.
> >
> > billing increment, I've seen a patch that implements this.  I was gonna
> >  port that forward to current rev.  If Darren or someone else doesn't get
> >  it first I'll get to it eventually.
> >
> > init_block - would implement at same time as billing_increment.
> >
> > connection_fee - again, like init_block.
> >
> > In fact, you *could* get all the values you need via the version of
> mod_lcr
> > that is sitting locally since you can query/set arbitrary channel vars
> via
> > custom sql.  You can decide how these fields relate to your lcr tables
> >  (they may not, it is up to you based on the sql you craft).
> >
> > You also mentioned you don't want to bill in real-time but only at end of
> > call.  This you can do today by turning off the heartbeat.   I think this
> >  is already discussed on the wiki.
> >
> > As always, a bounty goes a long way.   One of the devs might jump on it
> > sooner if there is money in it.  I'm swamped for the next week at least
> > so...  won't  have a chance to even look at this until then.  Not to
> >  mention I want to finish cleaning up my limit stuff.
> >
> > On Fri, Mar 26, 2010 at 6:12 PM, Jean-Yves F. Barbier
> <12ukwn at gmail.com>wrote:
> > > Le Fri, 26 Mar 2010 17:29:57 -0500,
> > > Anthony Minessale <anthony.minessale at gmail.com> a ?crit :
> > >
> > > Ok, thanks, I read the mod_nibblebill page and What I'd like to ask is
> to
> > > have some more features:
> > >
> > > * Connection fee
> > >
> > > * Time-credit (from french term "cr?dit-temps", means when you make a
> > >  call you're first invoiced a minimum of X seconds - sometimes along a
> > >  connection fee:(
> > >
> > > * A per second invoicing instead of per minute, but without a DB call
> > > every second, just one call at hangup.
> > >
> > > > you can post ideas here for discussion.
> > > > if you want something specific you can open a jira incident at
> > > > http://jira.freeswitch.org under "improvement" category.
> > >
> > > --
> > > Conscious is when you are aware of something and conscience is when you
> > > wish you weren't.
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From jason at jasonjgw.net  Fri Mar 26 20:56:11 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 27 Mar 2010 14:56:11 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
Message-ID: <20100327035611.GA3552@jdc.jasonjgw.net>

I am trying to test mod_silk with FreeSWITCH rev. 17118.

The problem appears to be that I haven't been able to get my FreeSWITCH to
offer SILK as one of its codecs.

mod_silk is loaded. I am using the PortAudio interface to make the call; the
other end is running a recently built FreeSWITCH as well, thanks to Bill Acker
for participating in the testing as always.

I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with
SILK at 24000h as the first item in the comma-separated list. I have also tried
it with SILK (no sampling rate specified).

The SIP trace doesn't show that it is being offered. Bill has configured the
other end in the same way.

For example:

a=rtpmap:10 L16/48000
a=rtpmap:114 CELT/48000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

is what appears as the SIP message from my local FreeSWITCH with
SILK at 24000h,CELT at 48000h,G722,BV32,PCMU,PCMA,GSM
in the codec preferences. (Actually, bv32 is out of date; I know the name
changed and I should fix it - I haven't been playing with these settings for a
while.)

Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and
IPv6 for making the connection.

I know Brian and others have this working, so there's obviously something
amiss here. As usual, there's no urgency - it's another attempt at a community
contribution through testing.



From mrene_lists at avgs.ca  Fri Mar 26 21:07:35 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 27 Mar 2010 00:07:35 -0400
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <1269651178.5486.102.camel@ktulu>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
	<1269651178.5486.102.camel@ktulu>
Message-ID: <F17F1609-D24C-47CC-BD59-661B81475028@avgs.ca>

And how exactly are you planning on detecting answer on the B-leg?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-03-26, at 8:52 PM, Victor Toofic wrote:

> Because the B leg is always immediately answered, it is giving false
> answer. The ringback, busy and fastbusy tones are being played once it
> is answered. And if I use the bridge app.. once the B leg gets  
> answered
> it answers the A leg.
>
> On Fri, 2010-03-26 at 19:33 -0500, Anthony Minessale wrote:
>> no,
>>
>> That sounds a little over-thought though?
>> Why can't you just use the bridge app?  We have extensive parameters
>> to do progress supervision on the wiki.
>>
>>
>> On Fri, Mar 26, 2010 at 7:12 PM, Victor Toofic <toofics at gmail.com>
>> wrote:
>>        I don't get it. What is "<dialstring2>" for? I'm not
>>        originating both
>>        legs of the call.
>>
>>        I already have the incoming A leg in the switch and I want it
>>        to stay in
>>        early media and "bridge" it one-way only to a originated B leg
>>        that is
>>        always answered since the very first second.
>>
>>        What I am trying to do is some kind of call progress
>>        supervision on the
>>        B leg and answer and bridge to the A leg only when I detect
>>        the B leg
>>        was really answered. Before that I just want to pass the audio
>>        coming
>>        from the answered B leg to the early-media-state A leg.. so
>>        the A leg
>>        can hear the ringback tones, voice-mail messages, etc.
>>
>>        I already detect the tones and when the B leg is really
>>        answered, at
>>        that moment I can do an uuid_bridge of both uuids and get the
>>        legs
>>        bridged. But I don't see how I could pass audio from B to A
>>        before
>>        answering the A leg.
>>
>>        Is this possible in any way with FS??
>>
>>        Maybe I didn't understand you, but I don't see how could it be
>>        possible
>>        to do it with an inline dialplan.
>>
>>        Thnks!!
>>
>>
>>
>>        On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
>>> yes,
>>>
>>> don't park it, have it dial the other leg with the bridge
>>        app.
>>>
>>> originate <dialstring1> bridge:<dialstring2> inline
>>>
>>>
>>> On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic
>>        <toofics at gmail.com>
>>> wrote:
>>>        Hi!
>>>
>>>        Does anybody know if there is a way to "bridge" an
>>        answered
>>>        channel to a
>>>        channel that is in "early media" state?
>>>
>>>        The scenario is like this:
>>>
>>>         Channel A               Channel B
>>>        -------------------> FS ------------->
>>>        (pre_answer, park)       (answer)
>>>        <---<---<---<---<--- Audio <---<---<---
>>>
>>>        When the channel A arrives I pre_answer and park it,
>>        so it is
>>>        in "early
>>>        media" state. After that I originate the channel B.
>>        When the
>>>        channel B
>>>        answers I just want to relay the audio coming from B
>>        to A, I
>>>        don't care
>>>        if A can't talk to B at that moment.
>>>
>>>        Is there a way to achieve this? Some ideas?
>>>
>>>        Thnks!
>>>
>>>        -
>>>        Regards,
>>>        V?ctor Toofic
>>>
>>>
>>>
>>>        _______________________________________________
>>>        FreeSWITCH-users mailing list
>>>        FreeSWITCH-users at lists.freeswitch.org
>>>
>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-users
>>>        http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>        _______________________________________________
>>        FreeSWITCH-users mailing list
>>        FreeSWITCH-users at lists.freeswitch.org
>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-users
>>        http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From toofics at gmail.com  Fri Mar 26 22:04:54 2010
From: toofics at gmail.com (Victor Toofic)
Date: Fri, 26 Mar 2010 23:04:54 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <F17F1609-D24C-47CC-BD59-661B81475028@avgs.ca>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
	<1269651178.5486.102.camel@ktulu>
	<F17F1609-D24C-47CC-BD59-661B81475028@avgs.ca>
Message-ID: <1269666294.5486.123.camel@ktulu>

Well, I've almost done that.

I can detect the ringback, fastbusy and voicemail tones, I can also
detect when the B-leg starts talking.

I am using tone_detect and VAD (in sip profile) to get these events and
act accordingly.

On Sat, 2010-03-27 at 00:07 -0400, Mathieu Rene wrote:
> And how exactly are you planning on detecting answer on the B-leg?
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-03-26, at 8:52 PM, Victor Toofic wrote:
> 
> > Because the B leg is always immediately answered, it is giving false
> > answer. The ringback, busy and fastbusy tones are being played once it
> > is answered. And if I use the bridge app.. once the B leg gets  
> > answered
> > it answers the A leg.
> >
> > On Fri, 2010-03-26 at 19:33 -0500, Anthony Minessale wrote:
> >> no,
> >>
> >> That sounds a little over-thought though?
> >> Why can't you just use the bridge app?  We have extensive parameters
> >> to do progress supervision on the wiki.
> >>
> >>
> >> On Fri, Mar 26, 2010 at 7:12 PM, Victor Toofic <toofics at gmail.com>
> >> wrote:
> >>        I don't get it. What is "<dialstring2>" for? I'm not
> >>        originating both
> >>        legs of the call.
> >>
> >>        I already have the incoming A leg in the switch and I want it
> >>        to stay in
> >>        early media and "bridge" it one-way only to a originated B leg
> >>        that is
> >>        always answered since the very first second.
> >>
> >>        What I am trying to do is some kind of call progress
> >>        supervision on the
> >>        B leg and answer and bridge to the A leg only when I detect
> >>        the B leg
> >>        was really answered. Before that I just want to pass the audio
> >>        coming
> >>        from the answered B leg to the early-media-state A leg.. so
> >>        the A leg
> >>        can hear the ringback tones, voice-mail messages, etc.
> >>
> >>        I already detect the tones and when the B leg is really
> >>        answered, at
> >>        that moment I can do an uuid_bridge of both uuids and get the
> >>        legs
> >>        bridged. But I don't see how I could pass audio from B to A
> >>        before
> >>        answering the A leg.
> >>
> >>        Is this possible in any way with FS??
> >>
> >>        Maybe I didn't understand you, but I don't see how could it be
> >>        possible
> >>        to do it with an inline dialplan.
> >>
> >>        Thnks!!
> >>
> >>
> >>
> >>        On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
> >>> yes,
> >>>
> >>> don't park it, have it dial the other leg with the bridge
> >>        app.
> >>>
> >>> originate <dialstring1> bridge:<dialstring2> inline
> >>>
> >>>
> >>> On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic
> >>        <toofics at gmail.com>
> >>> wrote:
> >>>        Hi!
> >>>
> >>>        Does anybody know if there is a way to "bridge" an
> >>        answered
> >>>        channel to a
> >>>        channel that is in "early media" state?
> >>>
> >>>        The scenario is like this:
> >>>
> >>>         Channel A               Channel B
> >>>        -------------------> FS ------------->
> >>>        (pre_answer, park)       (answer)
> >>>        <---<---<---<---<--- Audio <---<---<---
> >>>
> >>>        When the channel A arrives I pre_answer and park it,
> >>        so it is
> >>>        in "early
> >>>        media" state. After that I originate the channel B.
> >>        When the
> >>>        channel B
> >>>        answers I just want to relay the audio coming from B
> >>        to A, I
> >>>        don't care
> >>>        if A can't talk to B at that moment.
> >>>
> >>>        Is there a way to achieve this? Some ideas?
> >>>
> >>>        Thnks!
> >>>
> >>>        -
> >>>        Regards,
> >>>        V?ctor Toofic
> >>>
> >>>
> >>>
> >>>        _______________________________________________
> >>>        FreeSWITCH-users mailing list
> >>>        FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> >> freeswitch-users
> >>>        http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> FreeSWITCH http://www.freeswitch.org/
> >>> ClueCon http://www.cluecon.com/
> >>> Twitter: http://twitter.com/FreeSWITCH_wire
> >>>
> >>> AIM: anthm
> >>> MSN:anthony_minessale at hotmail.com
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org
> >>> googletalk:conf+888 at conference.freeswitch.org
> >>> pstn:+19193869900
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>>
> >>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> >> freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>        _______________________________________________
> >>        FreeSWITCH-users mailing list
> >>        FreeSWITCH-users at lists.freeswitch.org
> >>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> >> freeswitch-users
> >>        http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> -- 
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> >> users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> > users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lists at infosecurity.ch  Sat Mar 27 01:21:48 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Sat, 27 Mar 2010 09:21:48 +0100
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <1269666294.5486.123.camel@ktulu>
References: <1269632432.5486.54.camel@ktulu>	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>	<1269648734.5486.93.camel@ktulu>	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>	<1269651178.5486.102.camel@ktulu>	<F17F1609-D24C-47CC-BD59-661B81475028@avgs.ca>
	<1269666294.5486.123.camel@ktulu>
Message-ID: <4BADC01C.2020307@infosecurity.ch>

Wow!

When read can you share your solution on the wiki or on the mailing list?

Fabio
On 27/03/10 06.04, Victor Toofic wrote:
> Well, I've almost done that.
>
> I can detect the ringback, fastbusy and voicemail tones, I can also
> detect when the B-leg starts talking.
>
> I am using tone_detect and VAD (in sip profile) to get these events and
> act accordingly.
>
>    



From tculjaga at gmail.com  Sat Mar 27 01:50:27 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 27 Mar 2010 09:50:27 +0100
Subject: [Freeswitch-users] grandstream gxp1200
Message-ID: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>

hello guys,

anyone using that model ?

if yes, how do u do auto-provisioning ? ... I managed the phone to get the
config file during boot (via dhcp option 43) but it is not applying it ...
any hint ?
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From david.ponzone at gmail.com  Sat Mar 27 02:02:33 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 27 Mar 2010 10:02:33 +0100
Subject: [Freeswitch-users] session nb
In-Reply-To: <20100327040207.33ac6858@anubis.defcon1>
References: <20100327040207.33ac6858@anubis.defcon1>
Message-ID: <1ECA614A-B116-45CA-B494-7F430D882E7A@gmail.com>

Jean-Yves,

the max sessions and max cps in FS are arbitrary values, configured in  
switch.conf.xml.
All default installs of FS have the same.
The purpose of those values, that you are supposed to customize, is to  
protect FS against the worse (imagine you know you absolutely never  
exceed 20cps, if you set max cps to 30, you would protect yourself  
against burst of INVITEs from unknown locations/attackers).

For sessions max, there is no theoratical way at the moment to predict  
how many calls a server would handle.
It depends on the CPU, on the architecture of the server, on the type  
of RTP-forwarding you do, the type and amount of transcoding you  
do,etc...
You will have to discover that by yourself, for instance by doing load  
tests with sipp.

It would be interesting if people share their own real-life results on  
the wiki.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 27/03/2010 ? 04:02, Jean-Yves F. Barbier a ?crit :

> Hi list,
>
> On my old athlonXP-2600+, FS sessions max is 1000 & session limit is  
> 30
> (I guess connection per second?.)
>
> How does it translate in the real world?
> Is it going down a lot if I've got many transcoding channels (let's  
> say
> iLBC internal transcoding to G711 external)
> I know it couldn't be a sharp answer, but I need to have an idea of  
> what is
> possible (or not.)
>
> -- 
> This fortune is false.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Sat Mar 27 02:19:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 27 Mar 2010 10:19:21 +0100
Subject: [Freeswitch-users] suggestions
In-Reply-To: <20100327001250.3f72bf73@anubis.defcon1>
References: <20100326225718.7902f354@anubis.defcon1>
	<191c3a031003261529r1dc400e2yba25a7f675d0cbe9@mail.gmail.com>
	<20100327001250.3f72bf73@anubis.defcon1>
Message-ID: <6F1E9AFA-5E58-4CC0-B0A0-B4CB7321B09A@gmail.com>

Le 27/03/2010 ? 00:12, Jean-Yves F. Barbier a ?crit :

> Le Fri, 26 Mar 2010 17:29:57 -0500,

> * A per second invoicing instead of per minute, but without a DB  
> call every
>  second, just one call at hangup.

With a DB call/update only at hangup, you would not be able to detect  
in the DB that a user has depleted his credit in order to disconnect  
it while on call.
Then, the way to go would be to check the DB before a new call so that  
FS gets the maximum allowed call duration for this call, based on the  
remaining credit and the rate of the called prefix, and so FS can set  
sched_hangup before bridging:
http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_sched_hangup

I am not sure mod_nibblebill implement that, but that would be easy to  
do.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.





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From nandy1925 at gmail.com  Sat Mar 27 03:10:10 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sat, 27 Mar 2010 18:10:10 +0800
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
Message-ID: <7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>

i'm using ht503 ata but the auto-provisioning should be the same. u need the
ff:
1. GS configuration tool
2. configuration template
3. TFTP/HTTP server

general steps:
1. get a copy of the raw config text file and modify the settings
2. run the GS java program to create the cfgxxxxxxxxxxxx where the
xxxxxxxxxxxx is the MAC address of the IP phone
3. place this file on the TFTP/HTTP server
4. set the IP address of the configuration server in the IP phone via keypad
5. reboot it

hope it helps.

On Sat, Mar 27, 2010 at 4:50 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello guys,
>
> anyone using that model ?
>
> if yes, how do u do auto-provisioning ? ... I managed the phone to get the
> config file during boot (via dhcp option 43) but it is not applying it ...
> any hint ?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From yivzhenko at mksat.net  Sat Mar 27 04:36:14 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Sat, 27 Mar 2010 13:36:14 +0200
Subject: [Freeswitch-users] Bad Video in latest SVN
In-Reply-To: <191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>
References: <201003261323.50280.yivzhenko@mksat.net>
	<191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>
Message-ID: <201003271336.14407.yivzhenko@mksat.net>

I'm only upgrade binary files of freeswitch in my production system. all 
config files not changed. When i notice this problem, i'm restore previous 
version(16013) and begin discover this problem on test system.

I run freeswitch with default configuration, only change this strings in 
vars.xml: 
 <X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=PCMA,H264,H263,H263-1998,H263-2000"/>
<X-PRE-PROCESS cmd="set" 
data="outbound_codec_prefs=PCMA,H264,H263,H263-1998,H263-2000"/>

Problem still exist.

There is pastebin log of call setup
http://pastebin.freeswitch.org/12557 (debug level)
http://pastebin.freeswitch.org/12558 (internal siptrace on)

On Saturday 27 March 2010 00:48:27 Anthony Minessale wrote:
> what exactly are you doing?
> can you provide the dialplan and log of the call you are having problems
> with.
> the more information the better
>
> On Fri, Mar 26, 2010 at 6:23 AM, Yuriy Ivzhenko <yivzhenko at mksat.net> wrote:
> > Hi all,
> >
> > After upgrade from SVN 16013 to latest SVN 17109
> > Video quality become very bad.
> > I'm use H264 video codec and Grandstream GXV3140 phones
> >
> > Anyone knows why this may happen?
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From ben at langfeld.co.uk  Sat Mar 27 04:38:20 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sat, 27 Mar 2010 11:38:20 +0000
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
Message-ID: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>

Hey guys,

I've not used mailing lists much in the past, and this is the first one i've
used that has been this busy. After having been subscribed to the list for
about a month now (and even on digest mode) i've been having my inbox
crammed with list emails. Many of the threads are informative and
interesting, but some are not at all relevant to me. This has led me to
wonder, what is the reason behind using a list like this in preference to a
more structured forum? The separation, better search facilities, greater
granularity in subscriptions, etc all sound like compelling reasons for a
switch to my mind. I presume, however, that there is a reason for sticking
with mailman and i'm interested to discover what that is.

Regards,
Ben Langfeld

Wave > Email
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From yivzhenko at mksat.net  Sat Mar 27 04:48:05 2010
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Sat, 27 Mar 2010 13:48:05 +0200
Subject: [Freeswitch-users] Bad Video in latest SVN
In-Reply-To: <191c3a031003261731u25b6ca11v3421f36111b9fc64@mail.gmail.com>
References: <201003261323.50280.yivzhenko@mksat.net>
	<191c3a031003261548o1623bba2hb86e7f251b96fe95@mail.gmail.com>
	<191c3a031003261731u25b6ca11v3421f36111b9fc64@mail.gmail.com>
Message-ID: <201003271348.05247.yivzhenko@mksat.net>

Sorry, I have not seen last letter
SVN 17118. 
All works fine! )
Many thanks

On Saturday 27 March 2010 02:31:00 Anthony Minessale wrote:
> Never mind, I'm too impatient.
> Can you update to latest trunk 17117 and see if it's better?
>
>
> On Fri, Mar 26, 2010 at 5:48 PM, Anthony Minessale <
>
> anthony.minessale at gmail.com> wrote:
> > what exactly are you doing?
> > can you provide the dialplan and log of the call you are having problems
> > with.
> > the more information the better
> >
> > On Fri, Mar 26, 2010 at 6:23 AM, Yuriy Ivzhenko 
<yivzhenko at mksat.net>wrote:
> >> Hi all,
> >>
> >> After upgrade from SVN 16013 to latest SVN 17109
> >> Video quality become very bad.
> >> I'm use H264 video codec and Grandstream GXV3140 phones
> >>
> >> Anyone knows why this may happen?
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessal
> >e at gmail.com> IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at con
> >ference.freeswitch.org> pstn:+19193869900



From william.suffill at gmail.com  Sat Mar 27 04:51:44 2010
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 27 Mar 2010 07:51:44 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
Message-ID: <6b65470d1003270451m1eb74f76n456bc30526156844@mail.gmail.com>

It's been brought up before. I don't recall exactly all the +/- discussed
but there are some alternatives for the user depending on their preference.
Digest can get a bit annoying in high volume. The list is archived in
various places (Gmane.org,nabble.com)

Nabble does provide a forum style interface to the current list if you
rather follow it that way.
http://n2.nabble.com/freeswitch-users-f2379917.html
"This forum is an archive for the mailing list: *
freeswitch-users at lists.freeswitch.org* (more
options<http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
Messages posted here will be sent to this mailing list."

-- W
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From 12ukwn at gmail.com  Sat Mar 27 06:32:20 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 14:32:20 +0100
Subject: [Freeswitch-users] session nb
In-Reply-To: <1ECA614A-B116-45CA-B494-7F430D882E7A@gmail.com>
References: <20100327040207.33ac6858@anubis.defcon1>
	<1ECA614A-B116-45CA-B494-7F430D882E7A@gmail.com>
Message-ID: <20100327143220.6e13b9a5@anubis.defcon1>

Le Sat, 27 Mar 2010 10:02:33 +0100,
David Ponzone <david.ponzone at gmail.com> a ?crit :

> Jean-Yves,
> 
> the max sessions and max cps in FS are arbitrary values, configured in  
> switch.conf.xml.
> All default installs of FS have the same.

Arghh, at this time I didn't greped the code (actually my son will do it:
he's the programer) - I thought it was the result of tests that FS shows
at start.

> The purpose of those values, that you are supposed to customize, is to  
> protect FS against the worse (imagine you know you absolutely never  
> exceed 20cps, if you set max cps to 30, you would protect yourself  
> against burst of INVITEs from unknown locations/attackers).
> 
> For sessions max, there is no theoratical way at the moment to predict  
> how many calls a server would handle.
> It depends on the CPU, on the architecture of the server, on the type  
> of RTP-forwarding you do, the type and amount of transcoding you  
> do,etc...
> You will have to discover that by yourself, for instance by doing load  
> tests with sipp.

Ok, I already find it (also found sipbomber) because I thought I would have
such an answer :)

> It would be interesting if people share their own real-life results on  
> the wiki.

Yep, definitely

Merci David

-- 
Don't drop acid -- take it pass/fail.
		-- Seen in a Ladies' Room at Harvard


From mkezys at gmail.com  Sat Mar 27 07:27:16 2010
From: mkezys at gmail.com (Mindaugas Kezys)
Date: Sat, 27 Mar 2010 16:27:16 +0200
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
Message-ID: <0f1c01cacdb9$9a488c10$ced9a430$@com>

Stay away from Portech, we spent many many hours trying to solve problems with them.

Tried many firmware versions, contacted support, nothing helped. Worked with 6 devices 372/374. (All in garbage now).

Main issues:

1. Registration ? not registering or loosing registration
2. From time to time stops responding at all only reboot helps
3. Broken SIP implementation, check example below and find 2 errors (bad IP and port):

Call-ID: 23998cbe2e49c50f2af7ec483eea4a2d at 141.250.214.2020
Contact: <sip:101 at 141.250.214.202:0064>
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 234

Also from Asterisk CLI (1 nonsense in fullcontact):

[Aug 14 14:08:04] DEBUG[28172]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE devices SET ipaddr = '141.250.214.202', port = '5070', regseconds = '1250251804', username = '180', fullcontact = 'sip:180 at 141.250.214.202:507070;rinstance=7a35f8e1f1ecf3fe' WHERE name = '180'

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: info at kolmisoft.com
URL: http://www.kolmisoft.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nandy Dagondon
Sent: Friday, March 26, 2010 10:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] GSM gateway recommendation

hi itamar,

tks. are u happy w/ portech?

-nandy
On Fri, Mar 26, 2010 at 5:01 PM, Itamar Reis Peixoto <itamar at ispbrasil.com.br> wrote:
On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon <gcd at i.ph> wrote:
> hello everybody,
>
> i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> brand/model you had and your go/no-go recommendation.
>
> tks a lot,
> nandy
>
portech ?

--
------------

Itamar Reis Peixoto

e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From lists at infosecurity.ch  Sat Mar 27 07:58:25 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Sat, 27 Mar 2010 15:58:25 +0100
Subject: [Freeswitch-users] Multiplexing multiple rtp relaying incoming port
Message-ID: <4BAE1D11.3030704@infosecurity.ch>

Hi all,

found our that UDP/53 port is prioritized on most mobile operators as 
DNS traffic.

It would be really effective to be able to use only port 53 udp for 
incoming RTP traffic of FS while acting as Media Relay.

I tried with:
<param name="rtp-start-port" value="53"/>
<param name="rtp-end-port" value="53"/>

But i got error:
2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:74 
Rounding odd start port 53 to 54
2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:78 
Rounding odd end port 53 to 52
2010-03-27 15:54:57.566241 [CRIT] sofia_glue.c:712 No RTP ports available!
2010-03-27 15:54:57.566241 [ERR] sofia_glue.c:1203 
sofia/internal/sip:546547426 at 109.112.56.148:39664 I/O Error

If FS would be able to support a mode where it can receive multiple RTP 
streams coming from different peers on the same local port it would be 
very useful for mobile voip environment.

Fabio


From kristian.kielhofner at gmail.com  Sat Mar 27 08:13:11 2010
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Sat, 27 Mar 2010 11:13:11 -0400
Subject: [Freeswitch-users] Multiplexing multiple rtp relaying incoming
	port
In-Reply-To: <4BAE1D11.3030704@infosecurity.ch>
References: <4BAE1D11.3030704@infosecurity.ch>
Message-ID: <4d15ff861003270813g2d19002fubfd963f13cb2ca3e@mail.gmail.com>

On Sat, Mar 27, 2010 at 10:58 AM, Fabio Pietrosanti (naif)
<lists at infosecurity.ch> wrote:
> Hi all,
>
> found our that UDP/53 port is prioritized on most mobile operators as
> DNS traffic.
>
> It would be really effective to be able to use only port 53 udp for
> incoming RTP traffic of FS while acting as Media Relay.
>
> I tried with:
> <param name="rtp-start-port" value="53"/>
> <param name="rtp-end-port" value="53"/>
>
> But i got error:
> 2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:74
> Rounding odd start port 53 to 54
> 2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:78
> Rounding odd end port 53 to 52
> 2010-03-27 15:54:57.566241 [CRIT] sofia_glue.c:712 No RTP ports available!
> 2010-03-27 15:54:57.566241 [ERR] sofia_glue.c:1203
> sofia/internal/sip:546547426 at 109.112.56.148:39664 I/O Error
>
> If FS would be able to support a mode where it can receive multiple RTP
> streams coming from different peers on the same local port it would be
> very useful for mobile voip environment.
>
> Fabio
>

  There are many things wrong with this but let's start with the most
obvious: it won't take much time for these mobile operators to
distinguish your port 53 RTP traffic from real DNS traffic.  Even with
the same port number they look nothing alike and it would be trivial
for any provider worth their salt to separate and classify the traffic
accordingly.  Nevermind the fact that you'd be attracting A LOT of
attention for yourself.

  Your RTP traffic (at a rate of around 50pps) would hammer the
network and potentially DoS other user's legitimate DNS traffic.
Don't get greedy.

P.S. - If you were interested in doing this from a research
perspective I'd recommend developing your own RTP proxy (perhaps using
RTPProxy or MediaProxy as a base) to accomplish your "multiplexing"
goals although I think you're up for a challenge - just look at the
performance implications for any IAX2 implementation and you'll see
what I am talking about.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From 12ukwn at gmail.com  Sat Mar 27 08:26:43 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 16:26:43 +0100
Subject: [Freeswitch-users] Multiplexing multiple rtp relaying incoming
 port
In-Reply-To: <4BAE1D11.3030704@infosecurity.ch>
References: <4BAE1D11.3030704@infosecurity.ch>
Message-ID: <20100327162643.6f3e8f2d@anubis.defcon1>

Le Sat, 27 Mar 2010 15:58:25 +0100,
"Fabio Pietrosanti (naif)" <lists at infosecurity.ch> a ?crit :

> found our that UDP/53 port is prioritized on most mobile operators as 
> DNS traffic.
> 
> It would be really effective to be able to use only port 53 udp for 
> incoming RTP traffic of FS while acting as Media Relay.

Terrible idea, as Kristian said.
Provider will get mad and might cut the whole trafic on this port, leaving
you almost sitting duck.
 
> I tried with:
> <param name="rtp-start-port" value="53"/>
> <param name="rtp-end-port" value="53"/>
> 
> But i got error:
> 2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:74 
> Rounding odd start port 53 to 54

A comm needs 2 RTP streams, most of (good) softphones forbid an odd port
number in order to keep things clear: they give you an even port, and the
next one is the RTP "return path" of the same comm.

-- 
Most men would never get laid if it weren't for the pity fuck.


From javieraristizabal at gmail.com  Sat Mar 27 08:38:34 2010
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Sat, 27 Mar 2010 10:38:34 -0500
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <0f1c01cacdb9$9a488c10$ced9a430$@com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
	<0f1c01cacdb9$9a488c10$ced9a430$@com>
Message-ID: <b6f11f301003270838q497e7532wbb15361721aee6e0@mail.gmail.com>

You can try GSMOpen with many old cellulars.. :D

I know this http://www.zed-3.com/web/EN/Peripherals_GS8.htm

-- 
-.-- --- / ... --- -.--

Javier Aristiz?bal
Senior Developer



On Sat, Mar 27, 2010 at 9:27 AM, Mindaugas Kezys <mkezys at gmail.com> wrote:

> Stay away from Portech, we spent many many hours trying to solve problems
> with them.
>
> Tried many firmware versions, contacted support, nothing helped. Worked
> with 6 devices 372/374. (All in garbage now).
>
> Main issues:
>
> 1. Registration ? not registering or loosing registration
> 2. From time to time stops responding at all only reboot helps
> 3. Broken SIP implementation, check example below and find 2 errors (bad IP
> and port):
>
> Call-ID: 23998cbe2e49c50f2af7ec483eea4a2d at 141.250.214.2020
> Contact: <sip:101 at 141.250.214.202:0064>
> CSeq: 102 INVITE
> Content-Type: application/sdp
> Content-Length: 234
>
> Also from Asterisk CLI (1 nonsense in fullcontact):
>
> [Aug 14 14:08:04] DEBUG[28172]: res_config_mysql.c:356 update_mysql: MySQL
> RealTime: Update SQL: UPDATE devices SET ipaddr = '141.250.214.202', port =
> '5070', regseconds = '1250251804', username = '180', fullcontact =
> 'sip:180 at 141.250.214.202:507070;rinstance=7a35f8e1f1ecf3fe' WHERE name =
> '180'
>
> Regards,
> Mindaugas Kezys
>
> Kolmisoft UAB
> VoIP Billing Solutions
> e-mail: info at kolmisoft.com
> URL: http://www.kolmisoft.com
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nandy Dagondon
> Sent: Friday, March 26, 2010 10:19 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] GSM gateway recommendation
>
> hi itamar,
>
> tks. are u happy w/ portech?
>
> -nandy
> On Fri, Mar 26, 2010 at 5:01 PM, Itamar Reis Peixoto <
> itamar at ispbrasil.com.br> wrote:
> On Fri, Mar 26, 2010 at 2:49 AM, Nandy Dagondon <gcd at i.ph> wrote:
> > hello everybody,
> >
> > i'm looking for a GSM SIP gateway to be interfaced w/ FS. pls share what
> > brand/model you had and your go/no-go recommendation.
> >
> > tks a lot,
> > nandy
> >
> portech ?
>
> --
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at infosecurity.ch  Sat Mar 27 08:44:32 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Sat, 27 Mar 2010 16:44:32 +0100
Subject: [Freeswitch-users] Multiplexing multiple rtp relaying incoming
 port
In-Reply-To: <4d15ff861003270813g2d19002fubfd963f13cb2ca3e@mail.gmail.com>
References: <4BAE1D11.3030704@infosecurity.ch>
	<4d15ff861003270813g2d19002fubfd963f13cb2ca3e@mail.gmail.com>
Message-ID: <4BAE27E0.7070505@infosecurity.ch>

On 27/03/10 16.13, Kristian Kielhofner wrote:
>    There are many things wrong with this but let's start with the most
> obvious: it won't take much time for these mobile operators to
> distinguish your port 53 RTP traffic from real DNS traffic.  Even with
> the same port number they look nothing alike and it would be trivial
> for any provider worth their salt to separate and classify the traffic
> accordingly.  Nevermind the fact that you'd be attracting A LOT of
> attention for yourself.
>
>    Your RTP traffic (at a rate of around 50pps) would hammer the
> network and potentially DoS other user's legitimate DNS traffic.
> Don't get greedy.
>    
Well, that's a point of the whole vision.
Mobile VoIP it's difficult and require a lot of hacks and optimization 
and still it's difficult.

You can't control the QoS policy of mobile operator but if you know that 
they optimize some port or protocol try to use it.
You need to bypass eventually present block by obfuscating RTP and SIP 
protocol.
You need to use TCP port for SIP that does not have low NAT timeout of 
idle session.
You need to use VAD, high ptime, reduce protocol overhead and employ 
important jitter buffer optimization on SIP UA in order to manage the 
difficulties in the burst based mobile networks.

And the war never end, i do security mobile crypto product and in my 
context those tools are important when there are important international 
political events.
During that periods even the full udp could be blocked by mobile 
operators, DPI technologies can be enhanced to provide more informations 
to intelligence services and in such context the only transport you will 
have is HTTP (not even HTTPS) requiring to make VoIP over HTTP (dirty, 
slow, drammatically inefficient transport... brrr!).

So the possibility to multiplex across a single UDP port it's just one 
part of the big picture where the mobile is a pain in the XXX in any 
network scenario you want to look at it.
> P.S. - If you were interested in doing this from a research
> perspective I'd recommend developing your own RTP proxy (perhaps using
> RTPProxy or MediaProxy as a base) to accomplish your "multiplexing"
> goals although I think you're up for a challenge - just look at the
> performance implications for any IAX2 implementation and you'll see
> what I am talking about.
>    
mmmm probably if requiring a third party apps approach an RTP over DNS, 
by encapsulating packets within a DNS tunnel, would be even more effective.

Also i am seeing that the TCP socket of FS and the congestion control of 
TCP under mobile network have to be finely tuned and modified in order 
to handle the mobile network IP-related issues (high packet loss, high 
latency, high burst of traffic) where the retransmission of TCP and the 
exponential backoff algorithms could be killers.

Probably a TCP proxy with PEP approach is required to really use TCP on 
mobile environment for always on connection and realtime signaling. FS 
could also introduce such kind of support to be suited for mobile clients.

Fabio


From david.ponzone at gmail.com  Sat Mar 27 08:55:13 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 27 Mar 2010 16:55:13 +0100
Subject: [Freeswitch-users] Multiplexing multiple rtp relaying incoming
	port
In-Reply-To: <20100327162643.6f3e8f2d@anubis.defcon1>
References: <4BAE1D11.3030704@infosecurity.ch>
	<20100327162643.6f3e8f2d@anubis.defcon1>
Message-ID: <871F0CC5-E3FF-4DAF-B5BC-7B35895D7AB9@gmail.com>

Le 27/03/2010 ? 16:26, Jean-Yves F. Barbier a ?crit :

> A comm needs 2 RTP streams, most of (good) softphones forbid an odd  
> port
> number in order to keep things clear: they give you an even port,  
> and the
> next one is the RTP "return path" of the same comm.

Actually, a comm is 2 RTP streams but they use the same port on both  
ends (endpoint A port X <------> endpoint B port Y>.
The next port (often the odd one) is "used" by RTCP, so let's say it's  
unused as RTCP is not really widely used, but another RTP stream  
should never use this port, as it could lead to some strange things....



From toofics at gmail.com  Sat Mar 27 08:56:36 2010
From: toofics at gmail.com (Victor Toofic)
Date: Sat, 27 Mar 2010 09:56:36 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <4BADC01C.2020307@infosecurity.ch>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
	<1269651178.5486.102.camel@ktulu>
	<F17F1609-D24C-47CC-BD59-661B81475028@avgs.ca>
	<1269666294.5486.123.camel@ktulu>  <4BADC01C.2020307@infosecurity.ch>
Message-ID: <1269705396.5486.133.camel@ktulu>

Sure!

Im just working on the final details, pending with submiting a patch and
hoping it will get merged :D

On Sat, 2010-03-27 at 09:21 +0100, Fabio Pietrosanti (naif) wrote:
> Wow!
> 
> When read can you share your solution on the wiki or on the mailing list?
> 
> Fabio
> On 27/03/10 06.04, Victor Toofic wrote:
> > Well, I've almost done that.
> >
> > I can detect the ringback, fastbusy and voicemail tones, I can also
> > detect when the B-leg starts talking.
> >
> > I am using tone_detect and VAD (in sip profile) to get these events and
> > act accordingly.
> >
> >    
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sat Mar 27 09:15:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Mar 2010 11:15:14 -0500
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
Message-ID: <09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>

Because the rpm's were not compiled with -fPIC its getting fixed.. till then you'll have to compile it yourself.

/b

On Mar 26, 2010, at 10:16 PM, Neil Patel wrote:

> I can't get luasql to compile on CentOS 5.4:



From brian at freeswitch.org  Sat Mar 27 09:15:56 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Mar 2010 11:15:56 -0500
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <20100327035611.GA3552@jdc.jasonjgw.net>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
Message-ID: <FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>

Remove CELT.

/b

On Mar 26, 2010, at 10:56 PM, Jason White wrote:

> I am trying to test mod_silk with FreeSWITCH rev. 17118.
> 
> The problem appears to be that I haven't been able to get my FreeSWITCH to
> offer SILK as one of its codecs.
> 
> mod_silk is loaded. I am using the PortAudio interface to make the call; the
> other end is running a recently built FreeSWITCH as well, thanks to Bill Acker
> for participating in the testing as always.
> 
> I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with
> SILK at 24000h as the first item in the comma-separated list. I have also tried
> it with SILK (no sampling rate specified).
> 
> The SIP trace doesn't show that it is being offered. Bill has configured the
> other end in the same way.
> 
> For example:
> 
> a=rtpmap:10 L16/48000
> a=rtpmap:114 CELT/48000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> 
> is what appears as the SIP message from my local FreeSWITCH with
> SILK at 24000h,CELT at 48000h,G722,BV32,PCMU,PCMA,GSM
> in the codec preferences. (Actually, bv32 is out of date; I know the name
> changed and I should fix it - I haven't been playing with these settings for a
> while.)
> 
> Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and
> IPv6 for making the connection.
> 
> I know Brian and others have this working, so there's obviously something
> amiss here. As usual, there's no urgency - it's another attempt at a community
> contribution through testing.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lloyd.aloysius at gmail.com  Sat Mar 27 09:32:22 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sat, 27 Mar 2010 12:32:22 -0400
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
Message-ID: <8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>

Hi Niel,

Lua Mysql driver is not recommended. Use the Lua ODBC driver for the
connectivity. Enable the ODBC in the config file

Here is the modification need in the config file

CFLAGS= -O2 *-fPIC* $(WARN) -I$(COMPAT_DIR) $(DRIVER_INCS) $(INCS) $(DEFS)

Hope this will help you to solve the problem.

Thanks
Lloyd


On Sat, Mar 27, 2010 at 12:15 PM, Brian West <brian at freeswitch.org> wrote:

> Because the rpm's were not compiled with -fPIC its getting fixed.. till
> then you'll have to compile it yourself.
>
> /b
>
> On Mar 26, 2010, at 10:16 PM, Neil Patel wrote:
>
> > I can't get luasql to compile on CentOS 5.4:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Sat Mar 27 10:20:22 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 27 Mar 2010 18:20:22 +0100
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
Message-ID: <65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>

Hello Nandy,

this is exactly what i did i mean .. quite close...

i configured the DHCP server to pass the address of the http server to the
phone via DHCP Option 43 (vendor-specific) by tunneling option 66 within
option 43. With wireshark, i can see the phone gets the config file from the
specified server/path but for some reason it doesn't applay it.

do i need to configure something special on that template ? I've changed sip
server/accounts settings and some other things but it doesn't seem to be
working.

I got the template from
http://www.grandstream.com/support/configurationtool.html

do i need to encode it in a certan way before putting it to a http server?



On Sat, Mar 27, 2010 at 11:10 AM, Nandy Dagondon <nandy1925 at gmail.com>wrote:

> i'm using ht503 ata but the auto-provisioning should be the same. u need
> the ff:
> 1. GS configuration tool
> 2. configuration template
> 3. TFTP/HTTP server
>
> general steps:
> 1. get a copy of the raw config text file and modify the settings
> 2. run the GS java program to create the cfgxxxxxxxxxxxx where the
> xxxxxxxxxxxx is the MAC address of the IP phone
> 3. place this file on the TFTP/HTTP server
> 4. set the IP address of the configuration server in the IP phone via
> keypad
> 5. reboot it
>
> hope it helps.
>
> On Sat, Mar 27, 2010 at 4:50 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello guys,
>>
>> anyone using that model ?
>>
>> if yes, how do u do auto-provisioning ? ... I managed the phone to get the
>> config file during boot (via dhcp option 43) but it is not applying it ...
>> any hint ?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From 12ukwn at gmail.com  Sat Mar 27 11:39:09 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 19:39:09 +0100
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
Message-ID: <20100327193909.6a0e60a8@anubis.defcon1>

Le Sat, 27 Mar 2010 18:20:22 +0100,
Tihomir Culjaga <tculjaga at gmail.com> a ?crit :

I won't answer to your question but I'll told you about my experience with
grandstream: I've got 2 HT488 that were stopped for 2 years; recently I
re-plugged them and wanted to upgrade to the latest firmware.
PB: not any HT488 firmware in any places at gs, a bunch of people that have
the same PB on forums, and not even a word of answer from gs (sales,
tech, ...)

So, it tells me one thing: these guys are BOH that only want to take your
money and don't care once its done.

You can oppose this behaviour to Linksys: even if you have to dig a bit,
you're still able to recover stable or beta firmware for old appliances that
works correctly

HTH

...
> I got the template from
> http://www.grandstream.com/support/configurationtool.html
> 
> do i need to encode it in a certan way before putting it to a http server?

-- 
Halt!!  Who goes there, friend or enema?


From dave at 3c.co.uk  Sat Mar 27 12:05:35 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 27 Mar 2010 19:05:35 +0000
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <20100327193909.6a0e60a8@anubis.defcon1>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<20100327193909.6a0e60a8@anubis.defcon1>
Message-ID: <1269716735.12574.20.camel@local.freepabx.com>

A quick Google took me here:
http://www.grandstream.com/BETATEST/HT488_496_386/

On the other hand, trying to get proper information as to how to
securely remotely provision Linksys phones is a task for Hercules;
Grandstream just make the tools available to all.

--Dave

> Le Sat, 27 Mar 2010 18:20:22 +0100,
> Tihomir Culjaga <tculjaga at gmail.com> a ?crit :
> 
> I won't answer to your question but I'll told you about my experience with
> grandstream: I've got 2 HT488 that were stopped for 2 years; recently I
> re-plugged them and wanted to upgrade to the latest firmware.
> PB: not any HT488 firmware in any places at gs, a bunch of people that have
> the same PB on forums, and not even a word of answer from gs (sales,
> tech, ...)
> 
> So, it tells me one thing: these guys are BOH that only want to take your
> money and don't care once its done.
> 
> You can oppose this behaviour to Linksys: even if you have to dig a bit,
> you're still able to recover stable or beta firmware for old appliances that
> works correctly
> 
> HTH
> 
> ...
> > I got the template from
> > http://www.grandstream.com/support/configurationtool.html
> > 
> > do i need to encode it in a certan way before putting it to a http server?
> 



From 12ukwn at gmail.com  Sat Mar 27 12:27:15 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 27 Mar 2010 20:27:15 +0100
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <1269716735.12574.20.camel@local.freepabx.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<20100327193909.6a0e60a8@anubis.defcon1>
	<1269716735.12574.20.camel@local.freepabx.com>
Message-ID: <20100327202715.699c382a@anubis.defcon1>

Le Sat, 27 Mar 2010 19:05:35 +0000,
David Knell <dave at 3c.co.uk> a ?crit :

> A quick Google took me here:
> http://www.grandstream.com/BETATEST/HT488_496_386/
> 
> On the other hand, trying to get proper information as to how to
> securely remotely provision Linksys phones is a task for Hercules;
> Grandstream just make the tools available to all.
> 
> --Dave

Quite amazing: I can ensure you that before the beginning of february 2010,
not any info was available from them (I spent a whole week seeking and
trying to ask gs people - still no answer.)
And my Linksys example was just to make a parallel in service matter.

But I'm not stubborn: I can reconsider (AFA this firmware is working
correctly, which wasn't the case for a bunch of them.)

At this time, the least critics I heard about IP phones was about SNOM
(if some people could confirm that, it would be good:)

-- 
"Never make any mistaeks."
(Anonymous, in a mail discussion about to a kernel bug report.)


From infos at madovsky.org  Sat Mar 27 12:47:38 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 27 Mar 2010 15:47:38 -0400
Subject: [Freeswitch-users] need legA talks to listen leg B
Message-ID: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705>

If I don't talk from the legA no sound can be listen.
do I need to set a special vars to listen legB as soon as it answers ?
I don't use any vad.

Thanks
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From tacvbo at tacvbo.net  Sat Mar 27 13:01:56 2010
From: tacvbo at tacvbo.net (tacvbo at tacvbo.net)
Date: Sat, 27 Mar 2010 14:01:56 -0600
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com> 
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
Message-ID: <d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>

My case is with a Sangoma A200 card without HWEC and 4 FXO ports.
I'm using FreeSWITCH trunk 17119; when echo cancel is active in DAHDI
2.2.1 using mg2 or oslec and zt.conf looks like this:

[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 0
rxgain => 0.0
txgain => 0.0

the call [http://pastebin.freeswitch.org/12559] completes but caller
hears echo, if I modify echo_cancel_level to something bigger than 0
{32..64..} the call [http://pastebin.freeswitch.org/12560]  can't
complete because openzap don't detect dialtone.

Any clue?

On Fri, Mar 19, 2010 at 13:00, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I had a look at the code and I discovered that the zt mod defaults to a echo
> can level of 64 internally.
> I did a patch to the code that should allow it to be explicitly off when you
> omit that param.
>
> If you could update to latest trunk and test it, it should be a real test of
> my theory.
>
> If you still have the problem the next step would be to add some debugging
> code to
> the ozmod_zt read and write functions to see if it's sending data up to the
> app.
>
> Maybe there are some new ioctls or something they added for interop with
> OSLEC that we don't know about.
>
>
>
> On Fri, Mar 19, 2010 at 1:21 PM, <Russell.Mosemann at cune.org> wrote:
>>
>> Anthony Minessale <anthony.minessale at gmail.com> said:
>>
>> > All OpenZAP does with digium cards is open the device and read
>> > and write from it. If you are getting no audio the problem is
>> > happening in the kernel before OpenZAP gets to see anything.
>>
>> That provides a starting point for checking things.
>>
>> > My suggestion is to look at /usr/local/freeswitch/conf/zt.conf
>> > there is a line
>> > echo_cancel_level => 64
>> >
>> > try erasing or commenting it out with a leading ;
>>
>> I did a quick test, and that didn't solve the problem. I'll have to look
>> into it further.
>>
>> > >From my recollection when you use OSLEC, you have to disable the
>> > dahdi/zaptel echo canceler.
>>
>> That has changed in DAHDI. There is no default echo canceler. The
>> specific echo canceler (if any) must be specified for each channel (or
>> range of channels). OSLEC is referenced like any other echo canceler. For
>> example,
>>
>> echocanceller=oslec,1-23
>>
>> > BTW, you got that card for $100 because it really is worth
>> > that much, maybe less ;)
>>
>> No arguments there. This is merely a toy system, and audio quality is not
>> a factor. We just want to demonstrate that this can be useful for us.
>> Based on past experience with Sangoma products and support, it would be
>> wonderful to have a card from them.
>>
>> --
>> Russell Mosemann
> --
> Anthony Minessale II


From Russell.Mosemann at cune.org  Sat Mar 27 13:14:41 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 27 Mar 2010 15:14:41 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri><191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
	<d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
Message-ID: <E46A15E6FAB94736869CD046DAF08721@cune.pri>

tacvbo at tacvbo.net wrote:

> echo_cancel_level => 0

Comment that line.

; echo_cancel_level => 0

--
Russell Mosemann



From Russell.Mosemann at cune.org  Sat Mar 27 13:17:03 2010
From: Russell.Mosemann at cune.org (Russell Mosemann)
Date: Sat, 27 Mar 2010 15:17:03 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri><191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
	<d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
Message-ID: <238C02FDB3704B60AEF0A94E125523B5@cune.pri>

tacvbo at tacvbo.net wrote:

> My case is with a Sangoma A200 card without HWEC and 4 FXO ports.
> I'm using FreeSWITCH trunk 17119; when echo cancel is active in DAHDI
> 2.2.1 using mg2 or oslec and zt.conf looks like this:

I read your message more carefully. Shouldn't you be using the Sangoma driver with the Sangoma card instead of DAHDI?

--
Russell Mosemann



From d at d-man.org  Sat Mar 27 13:21:07 2010
From: d at d-man.org (Darren Schreiber)
Date: Sat, 27 Mar 2010 13:21:07 -0700
Subject: [Freeswitch-users] Sunday,
 April 25th - FreeSWITCH Users Group Install Fest - San Francisco, CA
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F177EC81776@EXVMBX020-3.exch020.serverdata.net>

Hi folks,
                Just an update for ya... I will be hosting our first FreeSWITCH Users Group Install Fest on April 25th, 2010 in downtown San Francisco. I don't have a location for you yet, but please sign up if the time works for you so I can gauge how big a place I need to find.

                http://www.meetup.com/fsusers/ for details, which I'll copy for you here, too:


Hi folks!

Welcome telecomm enthusiasts, current Asterisk/FreeSWITCH users and anyone else who's interested in SIP, FreeSWITCH or any other VoIP services! This is an exciting time in VoIP - here's your chance to network with others and learn more about FreeSWITCH and what others are doing with it.

We'll be having our first official meet-up on Sunday, April 25th 2010 at 4:00pm at a location downtown (TBD). We'll show you:

    * Installing FreeSWITCH
    * Setting up devices, trunks and general settings via XML
    * Using at least two cool features in FreeSWITCH (vote:)
      - mod_skypiax - Skype Integration
      - mod_event_socket - FreeSWITCH event sockets
      - mod_tts - Text to Speech Engine
      - mod_sofia - Advanced sofia w/ trunking/failover/etc.

Please plan to bring a laptop so you can get the most from this experience with hands-on interaction between the organizer and your peers.

The first 45 minutes of the event will be structured (so please try to be on time) and lead by the organizer, after which we'll share amongst each other. You'll also have the chance to network with other individuals who are working on similar projects.

A location will announced shortly but let us first know if you can make the date!

This is an independent event and is not sponsored by any other company nor officially endorsed by FreeSWITCH. It's more like a local users group.

Thanks much, and welcome!
   - Darren Schreiber

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From brian at freeswitch.org  Sat Mar 27 13:30:58 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Mar 2010 15:30:58 -0500
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com>
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com>
	<d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com>
Message-ID: <EEAC4F0A-3B09-48F5-8026-BC3F56AD4525@freeswitch.org>

You don't have to use Dahdi with any Sangoma card in conjunction with OpenZAP... its able to use the native interface.

/b

On Mar 27, 2010, at 3:01 PM, tacvbo at tacvbo.net wrote:

> My case is with a Sangoma A200 card without HWEC and 4 FXO ports.
> I'm using FreeSWITCH trunk 17119; when echo cancel is active in DAHDI
> 2.2.1 using mg2 or oslec and zt.conf looks like this:



From juancarlosg6 at gmail.com  Sat Mar 27 14:55:53 2010
From: juancarlosg6 at gmail.com (Juan Carlos Gonzalez Cardona)
Date: Sat, 27 Mar 2010 16:55:53 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 280
In-Reply-To: <mailman.921.1269380292.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.921.1269380292.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <d71129941003271455m3e0bbe1exe579c1a166572ebc@mail.gmail.com>

Hi, the problem is time limit for call to some extensions, not for all.

2010/3/23 <freeswitch-users-request at lists.freeswitch.org>

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Time limit (Juan Carlos Gonzalez Cardona)
>   2. Re: Compile issue with openzap (Joseph L. Casale)
>   3. Re: Time limit (Rupa Schomaker)
>   4. Re: Compile issue with openzap (Brian West)
>   5. donating german prompts (Christian L?schenkohl)
>
>
> ---------- Mensaje reenviado ----------
> From: Juan Carlos Gonzalez Cardona <juancarlosg6 at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 23 Mar 2010 15:14:42 -0500
> Subject: [Freeswitch-users] Time limit
>
> Hi everyone, is posible use call limite for calls, to some extensiones?
>
> In Asterisk is possible creating custom context with set time limit to
> specific trunk, but i see that Freeswitch is more stable code, i need to
> know if is possible create groups for limit calls and assign extensiones to
> that group, if is possible, how i do that, thank?s.
>
> --
> Juan Carlos Gonzalez C.
>
>
> ---------- Mensaje reenviado ----------
> From: "Joseph L. Casale" <jcasale at activenetwerx.com>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Tue, 23 Mar 2010 21:01:20 +0000
> Subject: Re: [Freeswitch-users] Compile issue with openzap
> >What is the output of the build when you type make?
>
> Ok, I didn't have access to that machine today, but on another
> following my notes from previous builds, I got the same errors
> again.
>
> I presume I missing a dep (CentOS 5 x64) but I compared my notes
> with the wiki and they match.
>
> make (both stdout/stderr) are pastebin'ed:
> http://pastebin.freeswitch.org/12529
>
> Thanks!
> jlc
>
>
>
>
> ---------- Mensaje reenviado ----------
> From: Rupa Schomaker <rupa at rupa.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 23 Mar 2010 16:03:12 -0500
> Subject: Re: [Freeswitch-users] Time limit
> for time limit, look at sched_hangup or sched_transfer
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_transfer
>
> If you want to limit the number of calls, look at mod_limit:
> http://wiki.freeswitch.org/wiki/Mod_limit
>
> On Tue, Mar 23, 2010 at 3:14 PM, Juan Carlos Gonzalez Cardona <
> juancarlosg6 at gmail.com> wrote:
>
>>
>> Hi everyone, is posible use call limite for calls, to some extensiones?
>>
>> In Asterisk is possible creating custom context with set time limit to
>> specific trunk, but i see that Freeswitch is more stable code, i need to
>> know if is possible create groups for limit calls and assign extensiones to
>> that group, if is possible, how i do that, thank?s.
>>
>> --
>> Juan Carlos Gonzalez C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
>
> ---------- Mensaje reenviado ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 23 Mar 2010 16:13:12 -0500
> Subject: Re: [Freeswitch-users] Compile issue with openzap
> /usr/bin/ld: cannot find -ljpeg
>
> install libjpeg-devel and you should be set.. its not an OpenZAP compile
> issue you're simply failing to pay closer attention to the errors.
>
> /b
>
> On Mar 23, 2010, at 4:01 PM, Joseph L. Casale wrote:
>
> >> What is the output of the build when you type make?
> >
> > Ok, I didn't have access to that machine today, but on another
> > following my notes from previous builds, I got the same errors
> > again.
> >
> > I presume I missing a dep (CentOS 5 x64) but I compared my notes
> > with the wiki and they match.
> >
> > make (both stdout/stderr) are pastebin'ed:
> > http://pastebin.freeswitch.org/12529
> >
> > Thanks!
> > jlc
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
>
>
> ---------- Mensaje reenviado ----------
> From: "Christian L?schenkohl" <christian.loeschenkohl at xpirio.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 23 Mar 2010 22:38:10 +0100
> Subject: [Freeswitch-users] donating german prompts
> hello
>
> we are going to produce a full set of german audio prompts and donate them
> to this wonderful project.
> i have attached a prompt list with the phrases we are going to record.
>
> the source of this prompt list is the file docs/phrase/phrase_de.xml. i
> have also compared it with
> the english file under docs/phrase/phrase_en.xml an merged this two +
> translated what was needed/had
> not been present in the german langunge file from english to german.
>
> i would be happy if someone could give me feedback on this and at best
> review the phrase list in terms
> of correctness and completeness.
>
> br
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Juan Carlos Gonzalez C.
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From fs-list at communicatefreely.net  Sat Mar 27 15:10:05 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Sat, 27 Mar 2010 18:10:05 -0400
Subject: [Freeswitch-users] new users in directory
In-Reply-To: <332BD48DAA2D4E5398B80C1F8E6470EB@MOBILEE1705>
References: <332BD48DAA2D4E5398B80C1F8E6470EB@MOBILEE1705>
Message-ID: <4BAE823D.1010502@communicatefreely.net>

Yes, the XML isn't parsed until you do.

If you really want dynamic users, look into mod_xml_curl.  You can put all your users in a database
and have a server side script create the xml, or if you don't want to go quite that far, you could
just supply a flat file to the web server that you update by hand.

If it bothers you that you have to do a reload, I would seriously look into the curl/scripting
solution.  Not only do you not need a reload, but you can avoid having to edit directory, sip
profile, dialplan, and pretty much everything else that you might have to adjust from time to time.
 It really makes Moves Adds and Changes easy when you only have to edit a few fields.

That's my two cents anyway.

-Tim

Madovsky wrote:
> good evening,
>  
> is it really necessary to do a "reloadxml" everytime a new user is created ?
>  
> Thanks
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Sat Mar 27 15:54:15 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 27 Mar 2010 15:54:15 -0700
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <6b65470d1003270451m1eb74f76n456bc30526156844@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<6b65470d1003270451m1eb74f76n456bc30526156844@mail.gmail.com>
Message-ID: <36A8A880-36D1-4F3A-8067-592CB24AD546@freeswitch.org>

There *IS* a forum. A volunteer set it up a few years ago and promised  
to maintain it. The fact that you can't find it is proof positive that  
a forum just isn't the best choice for this particular online  
community. I'd say the nabble site is your best option.

-MC

Sent from my iPhone

On Mar 27, 2010, at 4:51 AM, William Suffill  
<william.suffill at gmail.com> wrote:

> It's been brought up before. I don't recall exactly all the +/-  
> discussed but there are some alternatives for the user depending on  
> their preference. Digest can get a bit annoying in high volume. The  
> list is archived in various places (Gmane.org,nabble.com)
>
> Nabble does provide a forum style interface to the current list if  
> you rather follow it that way.
> http://n2.nabble.com/freeswitch-users-f2379917.html
> "This forum is an archive for the mailing list: freeswitch-users at lists.freeswitch.org 
>  (more options) Messages posted here will be sent to this mailing  
> list."
>
> -- W
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From nandy1925 at gmail.com  Sat Mar 27 15:59:42 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 28 Mar 2010 06:59:42 +0800
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
Message-ID: <7d0bfd8c1003271559q5bca68b9p66efc11e389ee628@mail.gmail.com>

> do i need to encode it in a certan way before putting it to a http server?

do u mean, you just placed the edited configuration (text file) on the http
server? no, you need to install the Configuration Tool program. the program
needs the ff parameters:
1. MAC address
2. template text file

it should produce a binary cfgxxxxxxxxxxxx file. this is the one u must
place in your server.

btw, i'm not familiar w/ DHCP Option 43. so, i use the keypad to point the
IP phone to the config server. :-)

-nandy

On Sun, Mar 28, 2010 at 1:20 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello Nandy,
>
> this is exactly what i did i mean .. quite close...
>
> i configured the DHCP server to pass the address of the http server to the
> phone via DHCP Option 43 (vendor-specific) by tunneling option 66 within
> option 43. With wireshark, i can see the phone gets the config file from the
> specified server/path but for some reason it doesn't applay it.
>
> do i need to configure something special on that template ? I've changed
> sip server/accounts settings and some other things but it doesn't seem to be
> working.
>
> I got the template from
> http://www.grandstream.com/support/configurationtool.html
>
> do i need to encode it in a certan way before putting it to a http server?
>
>
>
>
> On Sat, Mar 27, 2010 at 11:10 AM, Nandy Dagondon <nandy1925 at gmail.com>wrote:
>
>> i'm using ht503 ata but the auto-provisioning should be the same. u need
>> the ff:
>> 1. GS configuration tool
>> 2. configuration template
>> 3. TFTP/HTTP server
>>
>> general steps:
>> 1. get a copy of the raw config text file and modify the settings
>> 2. run the GS java program to create the cfgxxxxxxxxxxxx where the
>> xxxxxxxxxxxx is the MAC address of the IP phone
>> 3. place this file on the TFTP/HTTP server
>> 4. set the IP address of the configuration server in the IP phone via
>> keypad
>> 5. reboot it
>>
>> hope it helps.
>>
>> On Sat, Mar 27, 2010 at 4:50 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello guys,
>>>
>>> anyone using that model ?
>>>
>>> if yes, how do u do auto-provisioning ? ... I managed the phone to get
>>> the config file during boot (via dhcp option 43) but it is not applying it
>>> ... any hint ?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sat Mar 27 16:22:15 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 10:22:15 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
Message-ID: <20100327232215.GA8287@jdc.jasonjgw.net>

Brian West <brian at freeswitch.org> wrote:
> Remove CELT.

Unfortunately it didn't help.

a=rtpmap:10 L16/48000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

with the codes specified as
SILK at 24000h,G722,PCMU,PCMA

Further suggestions are welcome.



From dave at 3c.co.uk  Sat Mar 27 16:46:22 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 27 Mar 2010 23:46:22 +0000
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <20100327202715.699c382a@anubis.defcon1>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<20100327193909.6a0e60a8@anubis.defcon1>
	<1269716735.12574.20.camel@local.freepabx.com>
	<20100327202715.699c382a@anubis.defcon1>
Message-ID: <1269733582.12574.24.camel@local.freepabx.com>


On Sat, 2010-03-27 at 20:27 +0100, Jean-Yves F. Barbier wrote:

> At this time, the least critics I heard about IP phones was about SNOM
> (if some people could confirm that, it would be good:)

Last had the pleasure of using one maybe two or three years ago - its
number auto-completion feature drove me nuts (dial 2002 to call someone
internally and, if you'd previously dialled 02072002123, it'd dial that
instead); I've no doubt you can turn it off, but I just returned to my
simple Grandstream.

--Dave



From nandy1925 at gmail.com  Sat Mar 27 16:54:10 2010
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 28 Mar 2010 07:54:10 +0800
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <d3ceea381003261441j12f86a05ld850f09aee1e6870@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>
	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>
	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>
	<d3ceea381003261441j12f86a05ld850f09aee1e6870@mail.gmail.com>
Message-ID: <7d0bfd8c1003271654nb2f5a7eq930b104a9c0d9c09@mail.gmail.com>

hello guys,

@itamar, after reading mindaugas experience, what model have u tried?
@mindaugas, which brand(s) works? i'm interested in max. of 4 ports only.

nandy :-)
On Sat, Mar 27, 2010 at 5:41 AM, Itamar Reis Peixoto <
itamar at ispbrasil.com.br> wrote:

> On Fri, Mar 26, 2010 at 5:19 PM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
> > hi itamar,
> >
> > tks. are u happy w/ portech?
> >
> > -nandy
>
> yes, very good.
>
>
>
> ------------
>
> Itamar Reis Peixoto
>
> e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
> skype: itamarjp
> icq: 81053601
> +55 11 4063 5033
> +55 34 3221 8599
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 12ukwn at gmail.com  Sat Mar 27 17:11:08 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sun, 28 Mar 2010 01:11:08 +0100
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <1269733582.12574.24.camel@local.freepabx.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<20100327193909.6a0e60a8@anubis.defcon1>
	<1269716735.12574.20.camel@local.freepabx.com>
	<20100327202715.699c382a@anubis.defcon1>
	<1269733582.12574.24.camel@local.freepabx.com>
Message-ID: <20100328011108.1eb9d4fd@anubis.defcon1>

Le Sat, 27 Mar 2010 23:46:22 +0000,
David Knell <dave at 3c.co.uk> a ?crit :

> 
> On Sat, 2010-03-27 at 20:27 +0100, Jean-Yves F. Barbier wrote:
> 
> > At this time, the least critics I heard about IP phones was about SNOM
> > (if some people could confirm that, it would be good:)
> 
> Last had the pleasure of using one maybe two or three years ago - its
> number auto-completion feature drove me nuts (dial 2002 to call someone
> internally and, if you'd previously dialled 02072002123, it'd dial that
> instead); I've no doubt you can turn it off, but I just returned to my
> simple Grandstream.

I see that this kind of appliance's the same as others: m$ malpractices,
such as taking users for alpha-testers, has spread a lot :(

-- 
All men have the right to wait in line.


From ben at langfeld.co.uk  Sat Mar 27 17:44:02 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 28 Mar 2010 00:44:02 +0000
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
Message-ID: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>

There may be a forum, but such a thing cannot be expected to succeed if
there's competition with an already established list. Such a thing would
have to replace the mailing list to succeed. I believe this would help the
FreeSWITCH effort amongst newer users and lead to much faster growth. The
barrier to entry of a mailing list is a tad too high for some since it
presents an impression, justified or not, that the list is where the elite
hang out and is off-limits for newbies, whereas with a tidier forum setup
that would allow for a "Newbies Corner" might avoid this. Mailing lists are
showing their age these days - as demonstrated by the difficulty in
responding to this on digest mode - I have to strip away every other thread
first.

I'm convinced that a full shift to newer, better organised and more friendly
technology would only be a good thing for the FreeSWITCH project. Does
anyone agree, or am I out on my own on this one?

Regards,
Ben Langfeld

Wave > Email


On Sun, Mar 28, 2010 at 12:11 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

>
>
> ---------- Forwarded message ----------
> From: Michael S Collins <msc at freeswitch.org>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Date: Sat, 27 Mar 2010 15:54:15 -0700
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
> There *IS* a forum. A volunteer set it up a few years ago and promised to
> maintain it. The fact that you can't find it is proof positive that a forum
> just isn't the best choice for this particular online community. I'd say the
> nabble site is your best option.
>
> -MC
>
> Sent from my iPhone
>
> On Mar 27, 2010, at 4:51 AM, William Suffill <william.suffill at gmail.com>
> wrote:
>
> It's been brought up before. I don't recall exactly all the +/- discussed
> but there are some alternatives for the user depending on their preference.
> Digest can get a bit annoying in high volume. The list is archived in
> various places (Gmane.org, <http://nabble.com>nabble.com)
>
> Nabble does provide a forum style interface to the current list if you
> rather follow it that way.
> <http://n2.nabble.com/freeswitch-users-f2379917.html>
> http://n2.nabble.com/freeswitch-users-f2379917.html
> "This forum is an archive for the mailing list: *<freeswitch-users at lists.freeswitch.org>
> freeswitch-users at lists.freeswitch.org* (more options<http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
> Messages posted here will be sent to this mailing list."
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at 3c.co.uk  Sat Mar 27 17:50:51 2010
From: dave at 3c.co.uk (David Knell)
Date: Sat, 27 Mar 2010 18:50:51 -0600
Subject: [Freeswitch-users] Big conferences
Message-ID: <1269737451.12574.32.camel@local.freepabx.com>

Hi -

I've seen the odd query on the list about building and scaling large
audio conferences.  Given that most callers will be listen-only, it's
possible to scale up by transmitting a single RTP stream to multiple
callers.

Here's some proof-of-concept code, which roughly halves the CPU usage
for a 200-user (plus one speaker) conference, written in that highest of
performance languages, Perl:
http://www.softivr.com/wiki/index.php/HP_Conferencing

--Dave



From jason at jasonjgw.net  Sat Mar 27 18:13:59 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 12:13:59 +1100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
References: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
Message-ID: <20100328011359.GA10414@jdc.jasonjgw.net>

Ben Langfeld <ben at langfeld.co.uk> wrote:
> There may be a forum, but such a thing cannot be expected to succeed if
> there's competition with an already established list. Such a thing would
> have to replace the mailing list to succeed. I believe this would help the
> FreeSWITCH effort amongst newer users and lead to much faster growth. The
> barrier to entry of a mailing list is a tad too high for some since it
> presents an impression, justified or not, that the list is where the elite
> hang out and is off-limits for newbies, whereas with a tidier forum setup
> that would allow for a "Newbies Corner" might avoid this. Mailing lists are
> showing their age these days - as demonstrated by the difficulty in
> responding to this on digest mode - I have to strip away every other thread
> first.

Experienced users of mailing lists don't use digest mode. They filter the
mailing list into its own mail folder and use features of their mail user
agent to read it efficiently, e.g., threaded display.

I support to the existence of a Web interface to the mailing list, but I would
strongly object to any attempt to replace the list with a "Web forum", for a
number of reasons. In particular, if this were to happen, I know that I would
find it difficult to participate, perhaps impossible.

With a mailing list, I have a choice of software with which to read it, and I
have chosen carefully based on my accessibility needs and personal preferences
so that I can read and interact with mail efficiently. The problem with a Web
forum is that I have no choice: I would have to use a Web-based interface to
read and post messages, moreover, a Web interface provided by the forum
itself. Unfortunately, Web interfaces can't match my customized mail
environment for productivity; some of them are highly inaccessible to people
with disabilities, and they don't offer the flexibility of choosing and
customizing one's own client software that mailing lists and NNTP newsgroups
provide. Web fora are slow and difficult for me to use, at best, and impossible at worst, and I doubt that I am alone here.

So, I'm in favour of a Web interface but strongly against anything that
wouldn't also allow mailing list participation or offer NNTP access. That is,
I'm in favour of diversity and opposed to excluding people by forcing everyone
to use a single user interface - one provided by forum software.



From infos at madovsky.org  Sat Mar 27 18:15:48 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 27 Mar 2010 21:15:48 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
References: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
Message-ID: <F91AE3D445A5423EB813CE70A62BF853@MOBILEE1705>

I agree, but after 2 months iregistered
I must to confess that I really like this kind of anarchy.
It helps newbies to go quickly in the mood, elite to work
their human feels and sometimes fresh energy and ideas can also help
the old elite to push some intensity...

Regards


  ----- Original Message ----- 
  From: Ben Langfeld 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 27, 2010 8:44 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?


  There may be a forum, but such a thing cannot be expected to succeed if there's competition with an already established list. Such a thing would have to replace the mailing list to succeed. I believe this would help the FreeSWITCH effort amongst newer users and lead to much faster growth. The barrier to entry of a mailing list is a tad too high for some since it presents an impression, justified or not, that the list is where the elite hang out and is off-limits for newbies, whereas with a tidier forum setup that would allow for a "Newbies Corner" might avoid this. Mailing lists are showing their age these days - as demonstrated by the difficulty in responding to this on digest mode - I have to strip away every other thread first.


  I'm convinced that a full shift to newer, better organised and more friendly technology would only be a good thing for the FreeSWITCH project. Does anyone agree, or am I out on my own on this one?


  Regards,
  Ben Langfeld

  Wave > Email



  On Sun, Mar 28, 2010 at 12:11 AM, <freeswitch-users-request at lists.freeswitch.org> wrote:



    ---------- Forwarded message ----------
    From: Michael S Collins <msc at freeswitch.org>
    To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
    Date: Sat, 27 Mar 2010 15:54:15 -0700
    Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?

    There *IS* a forum. A volunteer set it up a few years ago and promised to maintain it. The fact that you can't find it is proof positive that a forum just isn't the best choice for this particular online community. I'd say the nabble site is your best option. 


    -MC

    Sent from my iPhone

    On Mar 27, 2010, at 4:51 AM, William Suffill <william.suffill at gmail.com> wrote:


      It's been brought up before. I don't recall exactly all the +/- discussed but there are some alternatives for the user depending on their preference. Digest can get a bit annoying in high volume. The list is archived in various places (Gmane.org,nabble.com)

      Nabble does provide a forum style interface to the current list if you rather follow it that way.
      http://n2.nabble.com/freeswitch-users-f2379917.html
      "This forum is an archive for the mailing list: freeswitch-users at lists.freeswitch.org (more options) Messages posted here will be sent to this mailing list."

      -- W

      _______________________________________________
      FreeSWITCH-users mailing list
      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
      http://www.freeswitch.org


------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From wchao at yahoo.com  Sat Mar 27 18:30:59 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Sat, 27 Mar 2010 21:30:59 -0400 (EDT)
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <20100328011359.GA10414@jdc.jasonjgw.net>
References: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
	<20100328011359.GA10414@jdc.jasonjgw.net>
Message-ID: <alpine.LRH.2.00.1003272126260.1948@osgood.microoffice.com>

I agree pretty strongly with Jason's viewpoint, and for the very same 
reasons. Not against a web forum, but other options (such as web forum) 
ought to be additional as opposed to in place of. Mailing lists might be 
"showing their age", but that's because they work. Procmail + Alpine works 
great, or an NNTP client is also perfectly fine. For people who want web 
access, there are web interfaces (like Nabble).


-------------------------------------
Date: Sun, 28 Mar 2010 12:13:59 +1100
From: Jason White <jason at jasonjgw.net>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?

Ben Langfeld <ben at langfeld.co.uk> wrote:
> There may be a forum, but such a thing cannot be expected to succeed if
> there's competition with an already established list. Such a thing would
> have to replace the mailing list to succeed. I believe this would help the
> FreeSWITCH effort amongst newer users and lead to much faster growth. The
> barrier to entry of a mailing list is a tad too high for some since it
> presents an impression, justified or not, that the list is where the elite
> hang out and is off-limits for newbies, whereas with a tidier forum setup
> that would allow for a "Newbies Corner" might avoid this. Mailing lists are
> showing their age these days - as demonstrated by the difficulty in
> responding to this on digest mode - I have to strip away every other thread
> first.

Experienced users of mailing lists don't use digest mode. They filter the
mailing list into its own mail folder and use features of their mail user
agent to read it efficiently, e.g., threaded display.

I support to the existence of a Web interface to the mailing list, but I would
strongly object to any attempt to replace the list with a "Web forum", for a
number of reasons. In particular, if this were to happen, I know that I would
find it difficult to participate, perhaps impossible.

With a mailing list, I have a choice of software with which to read it, and I
have chosen carefully based on my accessibility needs and personal preferences
so that I can read and interact with mail efficiently. The problem with a Web
forum is that I have no choice: I would have to use a Web-based interface to
read and post messages, moreover, a Web interface provided by the forum
itself. Unfortunately, Web interfaces can't match my customized mail
environment for productivity; some of them are highly inaccessible to people
with disabilities, and they don't offer the flexibility of choosing and
customizing one's own client software that mailing lists and NNTP newsgroups
provide. Web fora are slow and difficult for me to use, at best, and impossible at worst, and I doubt that I am alone here.

So, I'm in favour of a Web interface but strongly against anything that
wouldn't also allow mailing list participation or offer NNTP access. That is,
I'm in favour of diversity and opposed to excluding people by forcing everyone
to use a single user interface - one provided by forum software.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From ben at langfeld.co.uk  Sat Mar 27 18:38:26 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 28 Mar 2010 02:38:26 +0100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
Message-ID: <fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>

"So, I'm in favour of a Web interface but strongly against anything that
wouldn't also allow mailing list participation or offer NNTP access. That
is,
I'm in favour of diversity and opposed to excluding people by forcing
everyone
to use a single user interface - one provided by forum software."

So forum software with a "Subscribe to all by email" option and/or similar
RSS feed? Would that cover all the bases?

Regards,
Ben Langfeld

Wave > Email


On Sun, Mar 28, 2010 at 2:31 AM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: FreeSWITCH Mailing List - Why not a forum? (Ben Langfeld)
>   2. Big conferences (David Knell)
>   3. Re: FreeSWITCH Mailing List - Why not a forum? (Jason White)
>   4. Re: FreeSWITCH Mailing List - Why not a forum? (Madovsky)
>   5. Re: FreeSWITCH Mailing List - Why not a forum? (Wellie Chao)
>
>
> ---------- Forwarded message ----------
> From: Ben Langfeld <ben at langfeld.co.uk>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 28 Mar 2010 00:44:02 +0000
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
> There may be a forum, but such a thing cannot be expected to succeed if
> there's competition with an already established list. Such a thing would
> have to replace the mailing list to succeed. I believe this would help the
> FreeSWITCH effort amongst newer users and lead to much faster growth. The
> barrier to entry of a mailing list is a tad too high for some since it
> presents an impression, justified or not, that the list is where the elite
> hang out and is off-limits for newbies, whereas with a tidier forum setup
> that would allow for a "Newbies Corner" might avoid this. Mailing lists are
> showing their age these days - as demonstrated by the difficulty in
> responding to this on digest mode - I have to strip away every other thread
> first.
>
> I'm convinced that a full shift to newer, better organised and more
> friendly technology would only be a good thing for the FreeSWITCH project.
> Does anyone agree, or am I out on my own on this one?
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
>
> On Sun, Mar 28, 2010 at 12:11 AM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>>
>> ---------- Forwarded message ----------
>> From: Michael S Collins <msc at freeswitch.org>
>> To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Date: Sat, 27 Mar 2010 15:54:15 -0700
>> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
>> There *IS* a forum. A volunteer set it up a few years ago and promised to
>> maintain it. The fact that you can't find it is proof positive that a forum
>> just isn't the best choice for this particular online community. I'd say the
>> nabble site is your best option.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 27, 2010, at 4:51 AM, William Suffill <william.suffill at gmail.com>
>> wrote:
>>
>> It's been brought up before. I don't recall exactly all the +/- discussed
>> but there are some alternatives for the user depending on their preference.
>> Digest can get a bit annoying in high volume. The list is archived in
>> various places (Gmane.org, <http://nabble.com>nabble.com)
>>
>> Nabble does provide a forum style interface to the current list if you
>> rather follow it that way.
>> <http://n2.nabble.com/freeswitch-users-f2379917.html>
>> http://n2.nabble.com/freeswitch-users-f2379917.html
>> "This forum is an archive for the mailing list: *<freeswitch-users at lists.freeswitch.org>
>> freeswitch-users at lists.freeswitch.org* (more options<http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
>> Messages posted here will be sent to this mailing list."
>>
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> ---------- Forwarded message ----------
> From: David Knell <dave at 3c.co.uk>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 27 Mar 2010 18:50:51 -0600
> Subject: [Freeswitch-users] Big conferences
> Hi -
>
> I've seen the odd query on the list about building and scaling large
> audio conferences.  Given that most callers will be listen-only, it's
> possible to scale up by transmitting a single RTP stream to multiple
> callers.
>
> Here's some proof-of-concept code, which roughly halves the CPU usage
> for a 200-user (plus one speaker) conference, written in that highest of
> performance languages, Perl:
> http://www.softivr.com/wiki/index.php/HP_Conferencing
>
> --Dave
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Jason White <jason at jasonjgw.net>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sun, 28 Mar 2010 12:13:59 +1100
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
> Ben Langfeld <ben at langfeld.co.uk> wrote:
> > There may be a forum, but such a thing cannot be expected to succeed if
> > there's competition with an already established list. Such a thing would
> > have to replace the mailing list to succeed. I believe this would help
> the
> > FreeSWITCH effort amongst newer users and lead to much faster growth. The
> > barrier to entry of a mailing list is a tad too high for some since it
> > presents an impression, justified or not, that the list is where the
> elite
> > hang out and is off-limits for newbies, whereas with a tidier forum setup
> > that would allow for a "Newbies Corner" might avoid this. Mailing lists
> are
> > showing their age these days - as demonstrated by the difficulty in
> > responding to this on digest mode - I have to strip away every other
> thread
> > first.
>
> Experienced users of mailing lists don't use digest mode. They filter the
> mailing list into its own mail folder and use features of their mail user
> agent to read it efficiently, e.g., threaded display.
>
> I support to the existence of a Web interface to the mailing list, but I
> would
> strongly object to any attempt to replace the list with a "Web forum", for
> a
> number of reasons. In particular, if this were to happen, I know that I
> would
> find it difficult to participate, perhaps impossible.
>
> With a mailing list, I have a choice of software with which to read it, and
> I
> have chosen carefully based on my accessibility needs and personal
> preferences
> so that I can read and interact with mail efficiently. The problem with a
> Web
> forum is that I have no choice: I would have to use a Web-based interface
> to
> read and post messages, moreover, a Web interface provided by the forum
> itself. Unfortunately, Web interfaces can't match my customized mail
> environment for productivity; some of them are highly inaccessible to
> people
> with disabilities, and they don't offer the flexibility of choosing and
> customizing one's own client software that mailing lists and NNTP
> newsgroups
> provide. Web fora are slow and difficult for me to use, at best, and
> impossible at worst, and I doubt that I am alone here.
>
> So, I'm in favour of a Web interface but strongly against anything that
> wouldn't also allow mailing list participation or offer NNTP access. That
> is,
> I'm in favour of diversity and opposed to excluding people by forcing
> everyone
> to use a single user interface - one provided by forum software.
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Madovsky" <infos at madovsky.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 27 Mar 2010 21:15:48 -0400
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
>  I agree, but after 2 months iregistered
> I must to confess that I really like this kind of anarchy.
> It helps newbies to go quickly in the mood, elite to work
> their human feels and sometimes fresh energy and ideas can also help
> the old elite to push some intensity...
>
> Regards
>
>
>
> ----- Original Message -----
> *From:* Ben Langfeld <ben at langfeld.co.uk>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Saturday, March 27, 2010 8:44 PM
> *Subject:* Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a
> forum?
>
> There may be a forum, but such a thing cannot be expected to succeed if
> there's competition with an already established list. Such a thing would
> have to replace the mailing list to succeed. I believe this would help the
> FreeSWITCH effort amongst newer users and lead to much faster growth. The
> barrier to entry of a mailing list is a tad too high for some since it
> presents an impression, justified or not, that the list is where the elite
> hang out and is off-limits for newbies, whereas with a tidier forum setup
> that would allow for a "Newbies Corner" might avoid this. Mailing lists are
> showing their age these days - as demonstrated by the difficulty in
> responding to this on digest mode - I have to strip away every other thread
> first.
>
> I'm convinced that a full shift to newer, better organised and more
> friendly technology would only be a good thing for the FreeSWITCH project.
> Does anyone agree, or am I out on my own on this one?
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
>
> On Sun, Mar 28, 2010 at 12:11 AM, <
> freeswitch-users-request at lists.freeswitch.org> wrote:
>
>>
>>
>> ---------- Forwarded message ----------
>> From: Michael S Collins <msc at freeswitch.org>
>> To: "freeswitch-users at lists.freeswitch.org" <
>> freeswitch-users at lists.freeswitch.org>
>> Date: Sat, 27 Mar 2010 15:54:15 -0700
>> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
>>  There *IS* a forum. A volunteer set it up a few years ago and promised
>> to maintain it. The fact that you can't find it is proof positive that a
>> forum just isn't the best choice for this particular online community. I'd
>> say the nabble site is your best option.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Mar 27, 2010, at 4:51 AM, William Suffill <william.suffill at gmail.com>
>> wrote:
>>
>>  It's been brought up before. I don't recall exactly all the +/-
>> discussed but there are some alternatives for the user depending on their
>> preference. Digest can get a bit annoying in high volume. The list is
>> archived in various places (Gmane.org, <http://nabble.com>nabble.com)
>>
>> Nabble does provide a forum style interface to the current list if you
>> rather follow it that way.
>> <http://n2.nabble.com/freeswitch-users-f2379917.html>
>> http://n2.nabble.com/freeswitch-users-f2379917.html
>> "This forum is an archive for the mailing list: *<freeswitch-users at lists.freeswitch.org>
>> freeswitch-users at lists.freeswitch.org* (more options<http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
>> Messages posted here will be sent to this mailing list."
>>
>> -- W
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From: Wellie Chao <wchao at yahoo.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Sat, 27 Mar 2010 21:30:59 -0400 (EDT)
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
> I agree pretty strongly with Jason's viewpoint, and for the very same
> reasons. Not against a web forum, but other options (such as web forum)
> ought to be additional as opposed to in place of. Mailing lists might be
> "showing their age", but that's because they work. Procmail + Alpine works
> great, or an NNTP client is also perfectly fine. For people who want web
> access, there are web interfaces (like Nabble).
>
>
> -------------------------------------
> Date: Sun, 28 Mar 2010 12:13:59 +1100
> From: Jason White <jason at jasonjgw.net>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
>
> Ben Langfeld <ben at langfeld.co.uk> wrote:
> > There may be a forum, but such a thing cannot be expected to succeed if
> > there's competition with an already established list. Such a thing would
> > have to replace the mailing list to succeed. I believe this would help
> the
> > FreeSWITCH effort amongst newer users and lead to much faster growth. The
> > barrier to entry of a mailing list is a tad too high for some since it
> > presents an impression, justified or not, that the list is where the
> elite
> > hang out and is off-limits for newbies, whereas with a tidier forum setup
> > that would allow for a "Newbies Corner" might avoid this. Mailing lists
> are
> > showing their age these days - as demonstrated by the difficulty in
> > responding to this on digest mode - I have to strip away every other
> thread
> > first.
>
> Experienced users of mailing lists don't use digest mode. They filter the
> mailing list into its own mail folder and use features of their mail user
> agent to read it efficiently, e.g., threaded display.
>
> I support to the existence of a Web interface to the mailing list, but I
> would
> strongly object to any attempt to replace the list with a "Web forum", for
> a
> number of reasons. In particular, if this were to happen, I know that I
> would
> find it difficult to participate, perhaps impossible.
>
> With a mailing list, I have a choice of software with which to read it, and
> I
> have chosen carefully based on my accessibility needs and personal
> preferences
> so that I can read and interact with mail efficiently. The problem with a
> Web
> forum is that I have no choice: I would have to use a Web-based interface
> to
> read and post messages, moreover, a Web interface provided by the forum
> itself. Unfortunately, Web interfaces can't match my customized mail
> environment for productivity; some of them are highly inaccessible to
> people
> with disabilities, and they don't offer the flexibility of choosing and
> customizing one's own client software that mailing lists and NNTP
> newsgroups
> provide. Web fora are slow and difficult for me to use, at best, and
> impossible at worst, and I doubt that I am alone here.
>
> So, I'm in favour of a Web interface but strongly against anything that
> wouldn't also allow mailing list participation or offer NNTP access. That
> is,
> I'm in favour of diversity and opposed to excluding people by forcing
> everyone
> to use a single user interface - one provided by forum software.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Mar 27 18:55:12 2010
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 27 Mar 2010 20:55:12 -0500
Subject: [Freeswitch-users] Big conferences (OFF List reply)
In-Reply-To: <1269737451.12574.32.camel@local.freepabx.com>
Message-ID: <C7D42130.6A783%krice@freeswitch.org>

You know freeswitch conferencing already has this stuff built into C... You
just tag the users you don't want as speakers to be muted and it never calls
the mixer for those streams, it only mixes the talkers then muxes those
frames out to the other callers. Using this mechanism and an external call
director its possible to have 1000s of 'listeners' spread across multiple
boxes with talkers all confined to a 'master' box that also provides
shoutcasting etc


On 3/27/10 7:50 PM, "David Knell" <dave at 3c.co.uk> wrote:

> Hi -
> 
> I've seen the odd query on the list about building and scaling large
> audio conferences.  Given that most callers will be listen-only, it's
> possible to scale up by transmitting a single RTP stream to multiple
> callers.
> 
> Here's some proof-of-concept code, which roughly halves the CPU usage
> for a 200-user (plus one speaker) conference, written in that highest of
> performance languages, Perl:
> http://www.softivr.com/wiki/index.php/HP_Conferencing
> 
> --Dave
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Sat Mar 27 18:58:42 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 12:58:42 +1100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
References: <fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
Message-ID: <20100328015842.GA11026@jdc.jasonjgw.net>

Ben Langfeld <ben at langfeld.co.uk> wrote:
 
> So forum software with a "Subscribe to all by email" option and/or similar
> RSS feed? Would that cover all the bases?

No, not necessarily. An RSS feed wouldn't be enough; you can't use an RSS
client to post content, so it would have to be e-mail or NNTP to give users
choice of client, which is what I'm insisting on.

For e-mail support, it would have to handle the in-reply-to and references
headers properly to enable threading.

At this point we're talking about exactly what we have now: a mailing list
that can also be accessed on the Web. If you want NNTP access, this is also
available - see http://www.gmane.org/ or just refer your favourite NNTP news
reader to news.gmane.org and you'll find that you can read and (probably) post
to this list over NNTP too.

So my point is: we already have enough to satisfy everybody's preferences -
you can read and post via the Web, via e-mail or a newsgroup - we're not
excluding anybody on the basis of their needs and preferences regarding how to
participate - so the conclusion is that we should leave the mailing list in
place.

If you think there are shortcomings in the current Web interfaces then I'm
sure it would be possible to add another on top of the existing mailing list
setup.



From cmrienzo at gmail.com  Sat Mar 27 19:11:43 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Sat, 27 Mar 2010 22:11:43 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
Message-ID: <3687b0701003271911p38dd11e8m270a13db0b891273@mail.gmail.com>

http://old.nabble.com/Please-end-the-torment-td22352222.html
<http://old.nabble.com/Please-end-the-torment-td22352222.html%20>


On Sat, Mar 27, 2010 at 7:38 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Hey guys,
>
> I've not used mailing lists much in the past, and this is the first one
> i've used that has been this busy. After having been subscribed to the list
> for about a month now (and even on digest mode) i've been having my inbox
> crammed with list emails. Many of the threads are informative and
> interesting, but some are not at all relevant to me. This has led me to
> wonder, what is the reason behind using a list like this in preference to a
> more structured forum? The separation, better search facilities, greater
> granularity in subscriptions, etc all sound like compelling reasons for a
> switch to my mind. I presume, however, that there is a reason for sticking
> with mailman and i'm interested to discover what that is.
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sat Mar 27 19:31:21 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 13:31:21 +1100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <3687b0701003271911p38dd11e8m270a13db0b891273@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<3687b0701003271911p38dd11e8m270a13db0b891273@mail.gmail.com>
Message-ID: <20100328023121.GA11315@jdc.jasonjgw.net>

Also worth looking at:
http://dir.gmane.org/gmane.comp.telephony.freeswitch.user



From infos at madovsky.org  Sat Mar 27 19:57:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 27 Mar 2010 22:57:10 -0400
Subject: [Freeswitch-users] multiple continue_on_fail
Message-ID: <03E3E121FEFF45FC9A367250031C24B3@MOBILEE1705>

is it possible to have continue_on_fail set differently on after multiple bridges in an extension ?

Thanks
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From infos at madovsky.org  Sat Mar 27 19:58:59 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 27 Mar 2010 22:58:59 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<3687b0701003271911p38dd11e8m270a13db0b891273@mail.gmail.com>
Message-ID: <D8DD4AEBB9E143C5B5D89FEA7C553537@MOBILEE1705>

also

http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org

  ----- Original Message ----- 
  From: Christopher Rienzo 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, March 27, 2010 10:11 PM
  Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?


  http://old.nabble.com/Please-end-the-torment-td22352222.html 



  On Sat, Mar 27, 2010 at 7:38 AM, Ben Langfeld <ben at langfeld.co.uk> wrote:

    Hey guys,


    I've not used mailing lists much in the past, and this is the first one i've used that has been this busy. After having been subscribed to the list for about a month now (and even on digest mode) i've been having my inbox crammed with list emails. Many of the threads are informative and interesting, but some are not at all relevant to me. This has led me to wonder, what is the reason behind using a list like this in preference to a more structured forum? The separation, better search facilities, greater granularity in subscriptions, etc all sound like compelling reasons for a switch to my mind. I presume, however, that there is a reason for sticking with mailman and i'm interested to discover what that is.

    Regards,
    Ben Langfeld

    Wave > Email


    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    http://www.freeswitch.org






------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From tacvbo at tacvbo.net  Sat Mar 27 20:54:05 2010
From: tacvbo at tacvbo.net (tacvbo at tacvbo.net)
Date: Sat, 27 Mar 2010 21:54:05 -0600
Subject: [Freeswitch-users] freeswitch, openzap, dahdi issues...
In-Reply-To: <EEAC4F0A-3B09-48F5-8026-BC3F56AD4525@freeswitch.org>
References: <191c3a031003191042w4ce0ab40q9891cce0270b265b@mail.gmail.com> 
	<20100319182129.49EE5129EA7@cuneorg-email.cune.pri>
	<191c3a031003191200g40458c38rb40cec21771d04f8@mail.gmail.com> 
	<d9853e4f1003271301s1e302b50o388e2f37461c368f@mail.gmail.com> 
	<EEAC4F0A-3B09-48F5-8026-BC3F56AD4525@freeswitch.org>
Message-ID: <d9853e4f1003272054r4842bfd1le9dbf18e2f137353@mail.gmail.com>

On Sat, Mar 27, 2010 at 14:30, Brian West <brian at freeswitch.org> wrote:
> You don't have to use Dahdi with any Sangoma card in conjunction with OpenZAP... its able to use the native interface.

I know and I've tried that, I've forgot to mention: card do not have
HWEC. The problem is if I use TDM API mode, there is echo as if I use
DAHDI without SWEC.

Thanks for your responses.

-- 
Octavio H. Ruiz Cervera


From infos at madovsky.org  Sat Mar 27 21:07:51 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 28 Mar 2010 00:07:51 -0400
Subject: [Freeswitch-users] date format in FS log
Message-ID: <D91EF6572C8F4B6FB63CEE3BE6786EA0@MOBILEE1705>

Is there any way to add date in FS log
amd modify the format
like asterisk
 [general]
 dateformat=%F %Tbecause I'd like to create a filterwith fail2ban (an app that scan logs with a filter)to avoid any brute force attacks.there's an asterisk example herehttp://www.voip-info.org/wiki/index.php?page_id=5348I would like to test it with FS, and if it works well so why not wikiRegardsFranck
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From andrewkt at aktzero.com  Sun Mar 28 00:03:41 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Sun, 28 Mar 2010 03:03:41 -0400
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <20100327232215.GA8287@jdc.jasonjgw.net>
References: <20100327035611.GA3552@jdc.jasonjgw.net>	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
Message-ID: <4BAEFF4D.8030506@aktzero.com>

On 3/27/2010 7:22 PM, Jason White wrote:
> Brian West<brian at freeswitch.org>  wrote:
>    
>> Remove CELT.
>>      
> Unfortunately it didn't help.
>
> a=rtpmap:10 L16/48000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
>
> with the codes specified as
> SILK at 24000h,G722,PCMU,PCMA
>    
Try: silk at 8000h@20i,G722 at 20i,PCMU at 20i,PCMA at 20i

If it works, don't ask me why, just see: 
http://jira.freeswitch.org/browse/SFSIP-205
-- 

Andrew Thompson



From andrewkt at aktzero.com  Sun Mar 28 00:08:00 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Sun, 28 Mar 2010 03:08:00 -0400
Subject: [Freeswitch-users] need legA talks to listen leg B
In-Reply-To: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705>
References: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705>
Message-ID: <4BAF0050.6000308@aktzero.com>

On 3/27/2010 3:47 PM, Madovsky wrote:
> If I don't talk from the legA no sound can be listen.
> do I need to set a special vars to listen legB as soon as it answers ?
> I don't use any vad.

Watch the logs when you make one of these calls. See if when legA speaks 
the first time, there is immediately a green line in the log saying 
something about changing ports.

If it is, then look for 'comfort noise' settings on the device that is 
legA, could also be listed as a bandwidth preservation mechanism. Turn 
it off.

-- 
Andrew Thompson

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From jason at jasonjgw.net  Sun Mar 28 00:49:24 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 18:49:24 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <4BAEFF4D.8030506@aktzero.com>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
	<4BAEFF4D.8030506@aktzero.com>
Message-ID: <20100328074924.GA17282@jdc.jasonjgw.net>

Andrew Thompson <andrewkt at aktzero.com> wrote:
> On 3/27/2010 7:22 PM, Jason White wrote:
> > Brian West<brian at freeswitch.org>  wrote:
> >    
> >> Remove CELT.
> >>      
> > Unfortunately it didn't help.
> >
> > a=rtpmap:10 L16/48000
> > a=rtpmap:9 G722/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> >
> > with the codes specified as
> > SILK at 24000h,G722,PCMU,PCMA
> >    
> Try: silk at 8000h@20i,G722 at 20i,PCMU at 20i,PCMA at 20i

Thank you for the suggestion; no luck.

a=rtpmap:10 L16/48000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:10




From infos at madovsky.org  Sun Mar 28 01:03:56 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 28 Mar 2010 04:03:56 -0400
Subject: [Freeswitch-users] need legA talks to listen leg B
References: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705>
	<4BAF0050.6000308@aktzero.com>
Message-ID: <89203A70AD874529A0EA81B82D2C1E0A@MOBILEE1705>

with 
<action application="set" data="bridge_generate_comfort_noise=true"/> in profile
and
<action application="set" data="bridge_generate_comfort_noise=true"/> in dialplan
and
{bridge_generate_comfort_noise=true} in the bridge everything is ok

Thanks !

Franck
  ----- Original Message ----- 
  From: Andrew Thompson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 28, 2010 3:08 AM
  Subject: Re: [Freeswitch-users] need legA talks to listen leg B


  On 3/27/2010 3:47 PM, Madovsky wrote: 
    If I don't talk from the legA no sound can be listen.
    do I need to set a special vars to listen legB as soon as it answers ?
    I don't use any vad.

  Watch the logs when you make one of these calls. See if when legA speaks the first time, there is immediately a green line in the log saying something about changing ports.

  If it is, then look for 'comfort noise' settings on the device that is legA, could also be listed as a bandwidth preservation mechanism. Turn it off.


-- 
Andrew Thompson

------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From vetali100 at gmail.com  Sun Mar 28 01:24:37 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 28 Mar 2010 11:24:37 +0300
Subject: [Freeswitch-users]  Which module is responsible for directory
Message-ID: <a858e7691003280124k4c10b37cja0bee092f8bafb98@mail.gmail.com>

Hi,

I am using mod_xml_curl to get users from database and I don't need
./conf/directory/*.xml at all.
Is there is a module that I can unload (I would like to keep only needed
modules).

For example  the module responsible for diaplan is "mod_dialplan_xml".
Which one is for the users directory?

Thank you,
vIT
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From jason at jasonjgw.net  Sun Mar 28 02:17:58 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 20:17:58 +1100
Subject: [Freeswitch-users] Which module is responsible for directory
In-Reply-To: <a858e7691003280124k4c10b37cja0bee092f8bafb98@mail.gmail.com>
References: <a858e7691003280124k4c10b37cja0bee092f8bafb98@mail.gmail.com>
Message-ID: <20100328091758.GA17807@jdc.jasonjgw.net>

Vitalii Colosov <vetali100 at gmail.com> wrote:
> I am using mod_xml_curl to get users from database and I don't need
> ./conf/directory/*.xml at all.
> Is there is a module that I can unload (I would like to keep only needed
> modules).
> 
> For example  the module responsible for diaplan is "mod_dialplan_xml".
> Which one is for the users directory?

It appears to be in the core, so you can't unload it, and in any case it would
be the same code used to look up the directory entry irrespective of whether
the XML configuration is generated from files in a file system or database
queries. The directory/*.xml files aren't consulted once FreeSWITCH has
started; they're read only once to build the XML configuration in
log/freeswitch.xml.fsxml; with mod_XML_curl you're just searching the XML
dynamically via the database instead of reading from freeswitch.xml.fsxml.

So, there's nothing you can or need to do here.



From lists at infosecurity.ch  Sun Mar 28 02:44:45 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Sun, 28 Mar 2010 11:44:45 +0200
Subject: [Freeswitch-users] FS auto nat detection systems: are an attempt to
 provide alternative to STUN/ICE/TURN ?
Message-ID: <4BAF250D.5010405@infosecurity.ch>

Hi all,

i would like to ask one question to all FS folks.

I am really struggling against the automatic nat system of FS, where the 
full logic is still not really really clear and causing a lot of headheache.

Where on asterisk with a simple canreinvite=no everything works fine 
(even between dirty nat) on FS it does not work and FS start sending RTP 
to provide ip address on 10.0.0.0/8 (even if there is 
apply-nat-acl=rfc1918.auto) .

However, independently from the bugs of the automatic nat handling 
systems that have with the time to be noticed, discovered, fixed, etc, 
etc, my question is:

All this method for dynamic nat detection and handling of FS, want to be 
an alternative to the STUN/ICE/TURN approach of VoIP standard protocol?

Fabio


From jason at jasonjgw.net  Sun Mar 28 03:03:38 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 21:03:38 +1100
Subject: [Freeswitch-users] FS auto nat detection systems: are an
 attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <4BAF250D.5010405@infosecurity.ch>
References: <4BAF250D.5010405@infosecurity.ch>
Message-ID: <20100328100338.GA18005@jdc.jasonjgw.net>

Fabio Pietrosanti (naif) <lists at infosecurity.ch> wrote:
 
> All this method for dynamic nat detection and handling of FS, want to be 
> an alternative to the STUN/ICE/TURN approach of VoIP standard protocol?

I don't understand the question but you can choose in writing your
configuration how to handle NAT, including whether or not to use a STUN
server.



From mike at jerris.com  Sun Mar 28 03:03:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 28 Mar 2010 06:03:34 -0400
Subject: [Freeswitch-users] FS auto nat detection systems: are an
	attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <4BAF250D.5010405@infosecurity.ch>
References: <4BAF250D.5010405@infosecurity.ch>
Message-ID: <C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>

The situation you describe should not require anything not in the default configs of FreeSWITCH.  If its not working, please try reverting the configuration COMPLETELY back to the original configuration and it should work fine, then figure out what you changed to break it.


On Mar 28, 2010, at 5:44 AM, Fabio Pietrosanti (naif) wrote:
> I am really struggling against the automatic nat system of FS, where the 
> full logic is still not really really clear and causing a lot of headheache.
> 

No need to struggle, it works without changing anything.

> Where on asterisk with a simple canreinvite=no everything works fine 
> (even between dirty nat) on FS it does not work and FS start sending RTP 
> to provide ip address on 10.0.0.0/8 (even if there is 
> apply-nat-acl=rfc1918.auto) .
> 

What configuration have you changed?

> However, independently from the bugs of the automatic nat handling 
> systems that have with the time to be noticed, discovered, fixed, etc, 
> etc, my question is:
> 
> All this method for dynamic nat detection and handling of FS, want to be 
> an alternative to the STUN/ICE/TURN approach of VoIP standard protocol?

sometimes you simply can not avoid this approach, we have augmented it by throwing a little bit of upnp, natpmp, stun, and simple logic to get it right as much as it can.

Mike



From mike at jerris.com  Sun Mar 28 03:05:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 28 Mar 2010 06:05:15 -0400
Subject: [Freeswitch-users] date format in FS log
In-Reply-To: <D91EF6572C8F4B6FB63CEE3BE6786EA0@MOBILEE1705>
References: <D91EF6572C8F4B6FB63CEE3BE6786EA0@MOBILEE1705>
Message-ID: <3D9D0D24-689A-46C0-BAC1-96D8A6A212FB@jerris.com>

http://wiki.freeswitch.org/wiki/Fail2ban

^^^ check out this page that should help you out

On Mar 28, 2010, at 12:07 AM, Madovsky wrote:

> Is there any way to add date in FS log
> amd modify the format
> like asterisk
>  [general]
>  dateformat=%F %T
>  
> because I'd like to create a filter
> with fail2ban (an app that scan logs with a filter)
> to avoid any brute force attacks.
>  
> there's an asterisk example here
>  
> http://www.voip-info.org/wiki/index.php?page_id=5348
>  
> I would like to test it with FS, and if it works well so why not wiki

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From mike at jerris.com  Sun Mar 28 03:09:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 28 Mar 2010 06:09:40 -0400
Subject: [Freeswitch-users] multiple continue_on_fail
In-Reply-To: <03E3E121FEFF45FC9A367250031C24B3@MOBILEE1705>
References: <03E3E121FEFF45FC9A367250031C24B3@MOBILEE1705>
Message-ID: <57A4B542-D955-48F8-BB58-43C8482788B0@jerris.com>


On Mar 27, 2010, at 10:57 PM, Madovsky wrote:

> is it possible to have continue_on_fail set differently on after multiple bridges in an extension ?
>  


I could say yes, but maybe it would be better to say, its probably much quicker for you to every once in a while try something that would talk 30 seconds to find out before you send a stack of simple questions to the mailing list that you could have easily found out for yourself.  Your questions that can be simply answered by trying or typing a search phrase into the wiki to answer seem to be FAR FAR too frequent on the list.



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From jason at jasonjgw.net  Sun Mar 28 03:27:37 2010
From: jason at jasonjgw.net (Jason White)
Date: Sun, 28 Mar 2010 21:27:37 +1100
Subject: [Freeswitch-users] FS auto nat detection systems: are an
 attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>
References: <4BAF250D.5010405@infosecurity.ch>
	<C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>
Message-ID: <20100328102737.GA18174@jdc.jasonjgw.net>

Michael Jerris <mike at jerris.com> wrote:
> The situation you describe should not require anything not in the default
> configs of FreeSWITCH.  If its not working, please try reverting the
> configuration COMPLETELY back to the original configuration and it should
> work fine, then figure out what you changed to break it.

Also, there are helpful pages about NAT on the FreeSWITCh wiki.



From ben at langfeld.co.uk  Sun Mar 28 03:38:45 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Sun, 28 Mar 2010 02:38:45 -0800 (PST)
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <20100328015842.GA11026@jdc.jasonjgw.net>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
Message-ID: <1269772725049-4812504.post@n2.nabble.com>


"If you think there are shortcomings in the current Web interfaces then I'm
sure it would be possible to add another on top of the existing mailing list
setup."

Ok, so Nabble is a reasonable attempt but it still suffers from quoting
issues and a failure to strip off all of the rubbish mailman footers on
every post. It doesn't exactly scream minimalist and simple (DHH would have
a fit) and that's my issue. For now, though, it will have to do.

Could we at least make a modification to the FreeSWITCH site to make people
aware that these alternatives exist though? Having them hidden at crazy URLs
isn't helping anyone. Nabble is supposedly embeddable - can we stick it in
the main site and have it as prominent as the list?
-- 
View this message in context: http://n2.nabble.com/FreeSWITCH-Mailing-List-Why-not-a-forum-tp4808837p4812504.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dave at 3c.co.uk  Sun Mar 28 06:08:53 2010
From: dave at 3c.co.uk (David Knell)
Date: Sun, 28 Mar 2010 13:08:53 +0000
Subject: [Freeswitch-users] Big conferences (did Ken mean this to be
	an	OFF List reply?!)
In-Reply-To: <C7D42130.6A783%krice@freeswitch.org>
References: <C7D42130.6A783%krice@freeswitch.org>
Message-ID: <1269781733.12574.49.camel@local.freepabx.com>

Hi Ken,

Yes, to an extent.  But FS still has to transcode from L16 to whatever
codec each caller is using separately, run a thread for each caller,
etc., etc.

The test case which I ran used G.711 to minimise transcoding time, and
the 200 listeners didn't generate any RTP, which should have roughly the
same effect (looking at the code) as muting them in the conference.  I
suspect that there would be much more of a difference with a more
complex codec and/or listeners which generated RTP.

--Dave

> You know freeswitch conferencing already has this stuff built into C... You
> just tag the users you don't want as speakers to be muted and it never calls
> the mixer for those streams, it only mixes the talkers then muxes those
> frames out to the other callers. Using this mechanism and an external call
> director its possible to have 1000s of 'listeners' spread across multiple
> boxes with talkers all confined to a 'master' box that also provides
> shoutcasting etc
> 
> 
> On 3/27/10 7:50 PM, "David Knell" <dave at 3c.co.uk> wrote:
> 
> > Hi -
> > 
> > I've seen the odd query on the list about building and scaling large
> > audio conferences.  Given that most callers will be listen-only, it's
> > possible to scale up by transmitting a single RTP stream to multiple
> > callers.
> > 
> > Here's some proof-of-concept code, which roughly halves the CPU usage
> > for a 200-user (plus one speaker) conference, written in that highest of
> > performance languages, Perl:
> > http://www.softivr.com/wiki/index.php/HP_Conferencing
> > 
> > --Dave
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 



From stevendt at primrosebank.net  Sun Mar 28 08:08:27 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 28 Mar 2010 16:08:27 +0100
Subject: [Freeswitch-users] mod_skinny
Message-ID: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com>

Hi,

is mod_skinny (SCCP) still supported on FreeSWITCH ?

I see it in the Wiki, but it does not seem to be in any of the modules source directories.

regards
Dave
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From freeswitch-users-list at metik.com  Sun Mar 28 08:16:50 2010
From: freeswitch-users-list at metik.com (Metik)
Date: Sun, 28 Mar 2010 11:16:50 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
References: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
Message-ID: <4BAF72E2.5010907@metik.com>

There are many folks that are transient in terms of interests as 
evidenced by the serial nature of their endeavors and defunct projects 
(although I post occasionally, I prefer to contribute in other ways; 
e.g. occasional stray payments via paypal, helping users offline, 
etc.).  It does seem that a web forum, whether it is their ease of 
access or their churn in volunteer moderators or their proprietary 
software (which becomes a prison for content), generally reduces the 
quality of the information exchange.  Although I believe a better search 
interface would be a valuable tool to the mailing list as well as a a 
web-based forum (with RSS support, etc.) for "newbies", most users like 
the fact that one can use a program or service  (Thunderbird, Outlook, 
Gmail, etc.) that they are comfortable with to perform rather simple 
searches to watch topics in which they are interested.  All without 
having to worry about some poorly written SQL queries executed by a 
program running on under powered servers resting in the cloud that you 
have to go through layers (e.g. advanced search prompts, CAPTCHA, etc.) 
to do something that should only take a few seconds.  

What might be most useful is to author a more polished introduction 
(newbie) guide.  Although I don't have the time now to it myself, I 
would be willing to sponsor (i.e. fund) it so that more people are 
introduced to FS.

-metik


Ben Langfeld wrote:
> There may be a forum, but such a thing cannot be expected to succeed 
> if there's competition with an already established list. Such a thing 
> would have to replace the mailing list to succeed. I believe this 
> would help the FreeSWITCH effort amongst newer users and lead to much 
> faster growth. The barrier to entry of a mailing list is a tad too 
> high for some since it presents an impression, justified or not, that 
> the list is where the elite hang out and is off-limits for newbies, 
> whereas with a tidier forum setup that would allow for a "Newbies 
> Corner" might avoid this. Mailing lists are showing their age these 
> days - as demonstrated by the difficulty in responding to this on 
> digest mode - I have to strip away every other thread first.
>
> I'm convinced that a full shift to newer, better organised and more 
> friendly technology would only be a good thing for the FreeSWITCH 
> project. Does anyone agree, or am I out on my own on this one?
>
> Regards,
> Ben Langfeld
>
> Wave > Email
>
>
> On Sun, Mar 28, 2010 at 12:11 AM, 
> <freeswitch-users-request at lists.freeswitch.org 
> <mailto:freeswitch-users-request at lists.freeswitch.org>> wrote:
>
>
>
>     ---------- Forwarded message ----------
>     From: Michael S Collins <msc at freeswitch.org
>     <mailto:msc at freeswitch.org>>
>     To: "freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>"
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Date: Sat, 27 Mar 2010 15:54:15 -0700
>     Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not
>     a forum?
>     There *IS* a forum. A volunteer set it up a few years ago and
>     promised to maintain it. The fact that you can't find it is proof
>     positive that a forum just isn't the best choice for this
>     particular online community. I'd say the nabble site is your best
>     option. 
>
>     -MC
>
>     Sent from my iPhone
>
>     On Mar 27, 2010, at 4:51 AM, William Suffill
>     <william.suffill at gmail.com <mailto:william.suffill at gmail.com>> wrote:
>
>>     It's been brought up before. I don't recall exactly all the +/-
>>     discussed but there are some alternatives for the user depending
>>     on their preference. Digest can get a bit annoying in high
>>     volume. The list is archived in various places (Gmane.org
>>     <http://Gmane.org>,nabble.com <http://nabble.com>)
>>
>>     Nabble does provide a forum style interface to the current list
>>     if you rather follow it that way.
>>     http://n2.nabble.com/freeswitch-users-f2379917.html
>>     "This forum is an archive for the mailing list:
>>     *freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>* (more options
>>     <http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
>>     Messages posted here will be sent to this mailing list."
>>
>>     -- W
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From andrewkt at aktzero.com  Sun Mar 28 08:29:40 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Sun, 28 Mar 2010 11:29:40 -0400
Subject: [Freeswitch-users] mod_skinny
In-Reply-To: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com>
References: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com>
Message-ID: <4BAF75E4.7050100@aktzero.com>

On 3/28/2010 11:08 AM, Dave Stevenson wrote:
> Hi,
> is mod_skinny (SCCP) still supported on FreeSWITCH ?
> I see it in the Wiki, but it does not seem to be in any of the modules 
> source directories.
>
athompson at teros:~/freeswitch/src/mod/endpoints$ ls mod_skinny/
Makefile  Makefile.am  Makefile.in  mod_skinny.c  mod_skinny.h  
skinny_protocol.c  skinny_protocol.h  skinny_tables.c  skinny_tables.h  
test-skinny.pl

Seems all there to me. (I don't use it. But I seem to recall it getting 
some love recently.)

-- 
Andrew Thompson

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From andrewkt at aktzero.com  Sun Mar 28 08:40:33 2010
From: andrewkt at aktzero.com (Andrew Thompson)
Date: Sun, 28 Mar 2010 11:40:33 -0400
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <20100327232215.GA8287@jdc.jasonjgw.net>
References: <20100327035611.GA3552@jdc.jasonjgw.net>	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
Message-ID: <4BAF7871.3020909@aktzero.com>

On 3/27/2010 7:22 PM, Jason White wrote:
> Brian West<brian at freeswitch.org>  wrote:
>    
>> Remove CELT.
>>      
> Unfortunately it didn't help.
>    

Please pastebin a full call attempt with console logging at 7.

http://pastebin:freeswitch at pastebin.freeswitch.org/

-- 
Andrew Thompson



From stevendt at primrosebank.net  Sun Mar 28 09:01:54 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 28 Mar 2010 17:01:54 +0100
Subject: [Freeswitch-users] mod_skinny
References: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com>
	<4BAF75E4.7050100@aktzero.com>
Message-ID: <87ACE9B862C84831AD6667AEFAD31F2C@bp1.ad.bp.com>

Hi Andrew,

thanks for the reply, under Windows, I don't have a \endpoints\mod_skinny directory though.

Oooops - hold on - just downloaded the latest SVN (17126) and mod_skinny has just appeared !

As you say, there does seem to have been some recent work on it.

regards
Dave
  ----- Original Message ----- 
  From: Andrew Thompson 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, March 28, 2010 4:29 PM
  Subject: Re: [Freeswitch-users] mod_skinny


  On 3/28/2010 11:08 AM, Dave Stevenson wrote: 
    Hi,

    is mod_skinny (SCCP) still supported on FreeSWITCH ?

    I see it in the Wiki, but it does not seem to be in any of the modules source directories.


  athompson at teros:~/freeswitch/src/mod/endpoints$ ls mod_skinny/
  Makefile  Makefile.am  Makefile.in  mod_skinny.c  mod_skinny.h  skinny_protocol.c  skinny_protocol.h  skinny_tables.c  skinny_tables.h  test-skinny.pl

  Seems all there to me. (I don't use it. But I seem to recall it getting some love recently.)

-- 
Andrew Thompson

------------------------------------------------------------------------------


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From raffaele.p.guidi at gmail.com  Sun Mar 28 09:18:55 2010
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 28 Mar 2010 18:18:55 +0200
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <4BAF72E2.5010907@metik.com>
References: <fb4ab1e41003271744t3f96f2cdr50443c064a812077@mail.gmail.com>
	<4BAF72E2.5010907@metik.com>
Message-ID: <d5828a511003280918t7ffc4a5frc1095d83fe08d651@mail.gmail.com>

My 2 cents: I love this mailing list. I use gmail which has threading and -
even though I don't work on VoIP technologies anymore - I sometime spend
some minutes to look at these messages. So refreshing!

Ciao,
   R

On Sun, Mar 28, 2010 at 5:16 PM, Metik <freeswitch-users-list at metik.com>wrote:

> There are many folks that are transient in terms of interests as
> evidenced by the serial nature of their endeavors and defunct projects
> (although I post occasionally, I prefer to contribute in other ways;
> e.g. occasional stray payments via paypal, helping users offline,
> etc.).  It does seem that a web forum, whether it is their ease of
> access or their churn in volunteer moderators or their proprietary
> software (which becomes a prison for content), generally reduces the
> quality of the information exchange.  Although I believe a better search
> interface would be a valuable tool to the mailing list as well as a a
> web-based forum (with RSS support, etc.) for "newbies", most users like
> the fact that one can use a program or service  (Thunderbird, Outlook,
> Gmail, etc.) that they are comfortable with to perform rather simple
> searches to watch topics in which they are interested.  All without
> having to worry about some poorly written SQL queries executed by a
> program running on under powered servers resting in the cloud that you
> have to go through layers (e.g. advanced search prompts, CAPTCHA, etc.)
> to do something that should only take a few seconds.
>
> What might be most useful is to author a more polished introduction
> (newbie) guide.  Although I don't have the time now to it myself, I
> would be willing to sponsor (i.e. fund) it so that more people are
> introduced to FS.
>
> -metik
>
>
> Ben Langfeld wrote:
> > There may be a forum, but such a thing cannot be expected to succeed
> > if there's competition with an already established list. Such a thing
> > would have to replace the mailing list to succeed. I believe this
> > would help the FreeSWITCH effort amongst newer users and lead to much
> > faster growth. The barrier to entry of a mailing list is a tad too
> > high for some since it presents an impression, justified or not, that
> > the list is where the elite hang out and is off-limits for newbies,
> > whereas with a tidier forum setup that would allow for a "Newbies
> > Corner" might avoid this. Mailing lists are showing their age these
> > days - as demonstrated by the difficulty in responding to this on
> > digest mode - I have to strip away every other thread first.
> >
> > I'm convinced that a full shift to newer, better organised and more
> > friendly technology would only be a good thing for the FreeSWITCH
> > project. Does anyone agree, or am I out on my own on this one?
> >
> > Regards,
> > Ben Langfeld
> >
> > Wave > Email
> >
> >
> > On Sun, Mar 28, 2010 at 12:11 AM,
> > <freeswitch-users-request at lists.freeswitch.org
> > <mailto:freeswitch-users-request at lists.freeswitch.org>> wrote:
> >
> >
> >
> >     ---------- Forwarded message ----------
> >     From: Michael S Collins <msc at freeswitch.org
> >     <mailto:msc at freeswitch.org>>
> >     To: "freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>"
> >     <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     Date: Sat, 27 Mar 2010 15:54:15 -0700
> >     Subject: Re: [Freeswitch-users] FreeSWITCH Mailing List - Why not
> >     a forum?
> >     There *IS* a forum. A volunteer set it up a few years ago and
> >     promised to maintain it. The fact that you can't find it is proof
> >     positive that a forum just isn't the best choice for this
> >     particular online community. I'd say the nabble site is your best
> >     option.
> >
> >     -MC
> >
> >     Sent from my iPhone
> >
> >     On Mar 27, 2010, at 4:51 AM, William Suffill
> >     <william.suffill at gmail.com <mailto:william.suffill at gmail.com>>
> wrote:
> >
> >>     It's been brought up before. I don't recall exactly all the +/-
> >>     discussed but there are some alternatives for the user depending
> >>     on their preference. Digest can get a bit annoying in high
> >>     volume. The list is archived in various places (Gmane.org
> >>     <http://Gmane.org>,nabble.com <http://nabble.com>)
> >>
> >>     Nabble does provide a forum style interface to the current list
> >>     if you rather follow it that way.
> >>     http://n2.nabble.com/freeswitch-users-f2379917.html
> >>     "This forum is an archive for the mailing list:
> >>     *freeswitch-users at lists.freeswitch.org
> >>     <mailto:freeswitch-users at lists.freeswitch.org>* (more options
> >>     <
> http://n2.nabble.com/mailing_list/MailingListOptions.jtp?forum=2379917>)
> >>     Messages posted here will be sent to this mailing list."
> >>
> >>     -- W
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Mar 28 11:22:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 28 Mar 2010 13:22:28 -0500
Subject: [Freeswitch-users] Big conferences (did Ken mean this to be an
	OFF List reply?!)
In-Reply-To: <1269781733.12574.49.camel@local.freepabx.com>
References: <C7D42130.6A783%krice@freeswitch.org>
	<1269781733.12574.49.camel@local.freepabx.com>
Message-ID: <93769c21003281122q6b0018u27c92d2c79464f3d@mail.gmail.com>

the copy the encoded frame trick is nice, BUT it can't really be used
for any codec that has state.

On Sunday, March 28, 2010, David Knell <dave at 3c.co.uk> wrote:
> Hi Ken,
>
> Yes, to an extent. ?But FS still has to transcode from L16 to whatever
> codec each caller is using separately, run a thread for each caller,
> etc., etc.
>
> The test case which I ran used G.711 to minimise transcoding time, and
> the 200 listeners didn't generate any RTP, which should have roughly the
> same effect (looking at the code) as muting them in the conference. ?I
> suspect that there would be much more of a difference with a more
> complex codec and/or listeners which generated RTP.
>
> --Dave
>
>> You know freeswitch conferencing already has this stuff built into C... You
>> just tag the users you don't want as speakers to be muted and it never calls
>> the mixer for those streams, it only mixes the talkers then muxes those
>> frames out to the other callers. Using this mechanism and an external call
>> director its possible to have 1000s of 'listeners' spread across multiple
>> boxes with talkers all confined to a 'master' box that also provides
>> shoutcasting etc
>>
>>
>> On 3/27/10 7:50 PM, "David Knell" <dave at 3c.co.uk> wrote:
>>
>> > Hi -
>> >
>> > I've seen the odd query on the list about building and scaling large
>> > audio conferences. ?Given that most callers will be listen-only, it's
>> > possible to scale up by transmitting a single RTP stream to multiple
>> > callers.
>> >
>> > Here's some proof-of-concept code, which roughly halves the CPU usage
>> > for a 200-user (plus one speaker) conference, written in that highest of
>> > performance languages, Perl:
>> > http://www.softivr.com/wiki/index.php/HP_Conferencing
>>
>


From dave at 3c.co.uk  Sun Mar 28 12:05:13 2010
From: dave at 3c.co.uk (David Knell)
Date: Sun, 28 Mar 2010 19:05:13 +0000
Subject: [Freeswitch-users] Big conferences (did Ken mean this to be
	an	OFF List reply?!)
In-Reply-To: <93769c21003281122q6b0018u27c92d2c79464f3d@mail.gmail.com>
References: <C7D42130.6A783%krice@freeswitch.org>
	<1269781733.12574.49.camel@local.freepabx.com>
	<93769c21003281122q6b0018u27c92d2c79464f3d@mail.gmail.com>
Message-ID: <1269803113.12574.63.camel@local.freepabx.com>


On Sun, 2010-03-28 at 13:22 -0500, Michael Jerris wrote:
> the copy the encoded frame trick is nice, BUT it can't really be used
> for any codec that has state.

Not sure that this particularly affects things: any codec which retains
state from packet to packet still has to be able to recover from packet
loss, and so the decoder has somehow to converge towards the encoder's
state.  G.729 (just to take an example) typically takes 100ms for this
to happen - http://www.cs.columbia.edu/techreports/cucs-016-01.pdf - a
video stream would be corrupt until the next I-frame.

--Dave


> On Sunday, March 28, 2010, David Knell <dave at 3c.co.uk> wrote:
> > Hi Ken,
> >
> > Yes, to an extent.  But FS still has to transcode from L16 to whatever
> > codec each caller is using separately, run a thread for each caller,
> > etc., etc.
> >
> > The test case which I ran used G.711 to minimise transcoding time, and
> > the 200 listeners didn't generate any RTP, which should have roughly the
> > same effect (looking at the code) as muting them in the conference.  I
> > suspect that there would be much more of a difference with a more
> > complex codec and/or listeners which generated RTP.
> >
> > --Dave
> >
> >> You know freeswitch conferencing already has this stuff built into C... You
> >> just tag the users you don't want as speakers to be muted and it never calls
> >> the mixer for those streams, it only mixes the talkers then muxes those
> >> frames out to the other callers. Using this mechanism and an external call
> >> director its possible to have 1000s of 'listeners' spread across multiple
> >> boxes with talkers all confined to a 'master' box that also provides
> >> shoutcasting etc
> >>
> >>
> >> On 3/27/10 7:50 PM, "David Knell" <dave at 3c.co.uk> wrote:
> >>
> >> > Hi -
> >> >
> >> > I've seen the odd query on the list about building and scaling large
> >> > audio conferences.  Given that most callers will be listen-only, it's
> >> > possible to scale up by transmitting a single RTP stream to multiple
> >> > callers.
> >> >
> >> > Here's some proof-of-concept code, which roughly halves the CPU usage
> >> > for a 200-user (plus one speaker) conference, written in that highest of
> >> > performance languages, Perl:
> >> > http://www.softivr.com/wiki/index.php/HP_Conferencing
> >>
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jcasale at activenetwerx.com  Sun Mar 28 14:33:56 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 28 Mar 2010 21:33:56 +0000
Subject: [Freeswitch-users] Call Park issue
In-Reply-To: <191c3a031003261528t31c704x88a9e45504564b93@mail.gmail.com>
References: <CA5A491E9DEFBE4CB777DE97E21575E98EC2C0@prato.activenetwerx.local>
	<191c3a031003261457u2711122r4acd6154555a500e@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E98EC3BE@prato.activenetwerx.local>
	<191c3a031003261528t31c704x88a9e45504564b93@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E98EFE44@prato.activenetwerx.local>

>you should reproduce that with the console logging up
>
>enter this command at the CLI:
>
>console loglevel debug 
>
>then reproduce and post the results 

Anthony,
Looks like a phone config issue, I have to see what's different between the Aastra 480i's,
they aren't all working the same.

These phones have a built in Park function key, but require a server type followed
by the ext. Ever seen that setup to work w/ fs before?

Thanks,
jlc


From jason at jasonjgw.net  Sun Mar 28 20:39:49 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 29 Mar 2010 14:39:49 +1100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <1269772725049-4812504.post@n2.nabble.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
Message-ID: <20100329033949.GA19655@jdc.jasonjgw.net>

Ben Langfeld <ben at langfeld.co.uk> wrote:
> Could we at least make a modification to the FreeSWITCH site to make people
> aware that these alternatives exist though? Having them hidden at crazy URLs
> isn't helping anyone. Nabble is supposedly embeddable - can we stick it in
> the main site and have it as prominent as the list?

You could start a wiki page, if you think it would help, that documents all of
the means of accessing the mailing list. You could mention the IRC channel and
the FreeSWITCh conference there as well - in short, bring together on one page
all of the online fora available for the project.

What happens on the main site is up to the FreeSWITCH developers, but a wiki
page is open to community contributions.



From jason at jasonjgw.net  Sun Mar 28 21:01:46 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 29 Mar 2010 15:01:46 +1100
Subject: [Freeswitch-users] Trying to test mod_silk
In-Reply-To: <4BAF7871.3020909@aktzero.com>
References: <20100327035611.GA3552@jdc.jasonjgw.net>
	<FE824401-72E0-4FFB-BB56-3C99A727A877@freeswitch.org>
	<20100327232215.GA8287@jdc.jasonjgw.net>
	<4BAF7871.3020909@aktzero.com>
Message-ID: <20100329040146.GA19949@jdc.jasonjgw.net>

Andrew Thompson <andrewkt at aktzero.com> wrote:
 
> Please pastebin a full call attempt with console logging at 7.

http://pastebin.freeswitch.org/12565
with SILK at 24000h specified in my configuration as the only codec, and after
having verified that mod_silk was loaded. All of the SILK codecs (8000, 12000,
etc., including 24000) were logged as having been registered.

I didn't notice anything interesting in the logs that I've posted other than
the 48 khz codec being offered, presumably due to the fact that mod_portaudio
was configured for 48 khz. Note that this always works perfectly well with a
variety of codecs; FreeSWITCH resamples the input, as necessary, prior to
encoding it.



From vfclists at googlemail.com  Sun Mar 28 21:14:55 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 29 Mar 2010 04:14:55 +0000
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
Message-ID: <7abab2411003282114uc6e35bapbd9d0968b273e4e0@mail.gmail.com>

On 27 March 2010 11:38, Ben Langfeld <ben at langfeld.co.uk> wrote:
> Hey guys,
> I've not used mailing lists much in the past, and this is the first one i've
> used that has been this busy. After having been subscribed to the list for
> about a month now (and even on digest mode) i've been having my inbox
> crammed with list emails. Many of the threads are informative and
> interesting, but some are not at all relevant to me. This has led me to
> wonder, what is the reason behind using a list like this in preference to a
> more structured forum? The separation, better search facilities, greater
> granularity in subscriptions, etc all sound like compelling reasons for a
> switch to my mind. I presume, however, that there is a reason for sticking
> with mailman and i'm interested to discover what that is.
> Regards,
> Ben Langfeld
>
> Wave > Email
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

Warning: Mild Troll

forums are for wimps. real men use mailing lists.

Jokes aside, mailing lists are much better because all your interests
are centralized around your mail clients. When you subscribe to
different forums or mailing lists, you always have to be flitting
around from forum to forum to see if you've had responses. When I use
forums I tend to lose track of the enquires I have made.

Web browsing can be a great source of distraction, the urge to click
here and click there can be strong, with adverts and a some other
thingies vying to attract your atttention. With mailing lists you only
have to hear the sound of email delivery, look at the mail header and
decide whether to get back to work or attend to it. When you visit a
web page to look at a forum, you usually make a few more clicks which
are not related to your tasks.

It is only recently that I have focused on the idea of receiving
emails whenever a forum thread I am involved in receives a post that
forums have become more effective for me, which only shows how more
effective emails are. As some one said, set up your mail client to
filter on threads you post in and you should be okay. I haven't made
any queries on Freeswitch mailing lists for while, but just by using
email, I have noticed this thread and been able to post to it.

Mailing lists also mean that you are more likely to notice others
questions early and help with them even if you don't visit the forums
that much, and also know more of what is going on.

Effectiveness should take precedence over fashion and

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From neilp at cs.stanford.edu  Sun Mar 28 21:44:39 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sun, 28 Mar 2010 21:44:39 -0700
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com> 
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
Message-ID: <e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>

Hi Lloyd,

Even after the change to the config file below, I can't get the package to
compile:

[root at topaz luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
-Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
-DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
 src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
/usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol'
can not be used when making a shared object; recompile with -fPIC
src/luasql.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make: *** [src/odbc.so] Error 1

Anything I need to change or do other than add -fPIC to CFLAGS?

Thanks,
Neil

On Sat, Mar 27, 2010 at 9:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi Niel,
>
> Lua Mysql driver is not recommended. Use the Lua ODBC driver for the
> connectivity. Enable the ODBC in the config file
>
> Here is the modification need in the config file
>
> CFLAGS= -O2 *-fPIC* $(WARN) -I$(COMPAT_DIR) $(DRIVER_INCS) $(INCS) $(DEFS)
>
> Hope this will help you to solve the problem.
>
> Thanks
> Lloyd
>
>
> On Sat, Mar 27, 2010 at 12:15 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Because the rpm's were not compiled with -fPIC its getting fixed.. till
>> then you'll have to compile it yourself.
>>
>> /b
>>
>> On Mar 26, 2010, at 10:16 PM, Neil Patel wrote:
>>
>> > I can't get luasql to compile on CentOS 5.4:
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Sun Mar 28 21:58:28 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 29 Mar 2010 00:58:28 -0400
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
Message-ID: <08AEC5AE-F3B9-44D9-BD78-93BF737C478D@avgs.ca>

Try:

rm -f src/luasql.o
make


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2010-03-29, at 12:44 AM, Neil Patel wrote:

> Hi Lloyd,
>
> Even after the change to the config file below, I can't get the  
> package to compile:
>
> [root at topaz luasql-2.1.1]# make
> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall - 
> Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../ 
> compat/src -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o  
> src/odbc.so -shared  src/luasql.o src/ls_odbc.o  -L/usr/local/lib - 
> lodbc
> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local  
> symbol' can not be used when making a shared object; recompile with - 
> fPIC
> src/luasql.o: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> make: *** [src/odbc.so] Error 1
>
> Anything I need to change or do other than add -fPIC to CFLAGS?
>
> Thanks,
> Neil
>
> On Sat, Mar 27, 2010 at 9:32 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com 
> > wrote:
> Hi Niel,
>
> Lua Mysql driver is not recommended. Use the Lua ODBC driver for the  
> connectivity. Enable the ODBC in the config file
>
> Here is the modification need in the config file
>
> CFLAGS= -O2 -fPIC $(WARN) -I$(COMPAT_DIR) $(DRIVER_INCS) $(INCS) $ 
> (DEFS)
>
> Hope this will help you to solve the problem.
>
> Thanks
> Lloyd
>
>
> On Sat, Mar 27, 2010 at 12:15 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Because the rpm's were not compiled with -fPIC its getting fixed..  
> till then you'll have to compile it yourself.
>
> /b
>
> On Mar 26, 2010, at 10:16 PM, Neil Patel wrote:
>
> > I can't get luasql to compile on CentOS 5.4:
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Sun Mar 28 22:03:26 2010
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Mon, 29 Mar 2010 14:03:26 +0900
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
Message-ID: <15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>

On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi Lloyd,
>
> Even after the change to the config file below, I can't get the package to
> compile:
>
> [root at topaz luasql-2.1.1]# make
> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol'
> can not be used when making a shared object; recompile with -fPIC
> src/luasql.o: could not read symbols: Bad value
> collect2: ld returned 1 exit status
> make: *** [src/odbc.so] Error 1
>
> Anything I need to change or do other than add -fPIC to CFLAGS?
>

Here's
the script I use.
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From mkezys at gmail.com  Sun Mar 28 23:56:38 2010
From: mkezys at gmail.com (Mindaugas Kezys)
Date: Mon, 29 Mar 2010 09:56:38 +0300
Subject: [Freeswitch-users] GSM gateway recommendation
In-Reply-To: <7d0bfd8c1003271654nb2f5a7eq930b104a9c0d9c09@mail.gmail.com>
References: <7d0bfd8c1003252249k5d67a181jf3b634fc9816b216@mail.gmail.com>	<d3ceea381003260201k232afdbfob068ee86c2bb6957@mail.gmail.com>	<7d0bfd8c1003261319i22c92bar3e9730d8eba5370f@mail.gmail.com>	<d3ceea381003261441j12f86a05ld850f09aee1e6870@mail.gmail.com>
	<7d0bfd8c1003271654nb2f5a7eq930b104a9c0d9c09@mail.gmail.com>
Message-ID: <007e01cacf0c$fb34f730$f19ee590$@com>

2N VoiceBlue works fine for us with some modifications in configuration, such as: http://wiki.kolmisoft.com/index.php/VoiceBlue_Ring_problem

 

Regards,

Mindaugas Kezys

 

Kolmisoft UAB 

VoIP Billing Solutions

e-mail: info at kolmisoft.com

URL: http://www.kolmisoft.com

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nandy Dagondon
Sent: Sunday, March 28, 2010 1:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] GSM gateway recommendation

 

hello guys,

@itamar, after reading mindaugas experience, what model have u tried?
@mindaugas, which brand(s) works? i'm interested in max. of 4 ports only.

nandy :-)

On Sat, Mar 27, 2010 at 5:41 AM, Itamar Reis Peixoto <itamar at ispbrasil.com.br> wrote:

On Fri, Mar 26, 2010 at 5:19 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:
> hi itamar,
>
> tks. are u happy w/ portech?
>
> -nandy

yes, very good.




------------

Itamar Reis Peixoto

e-mail/msn/google talk/sip: itamar at ispbrasil.com.br
skype: itamarjp
icq: 81053601
+55 11 4063 5033
+55 34 3221 8599

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From neilp at cs.stanford.edu  Mon Mar 29 00:30:02 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 29 Mar 2010 00:30:02 -0700
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com> 
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com> 
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com> 
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>
Message-ID: <e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com>

Thanks all, got it to compile and seems like my odbc connection is working
properly:

[root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL>

In my lua script I have

require "luasql.odbc"

But am getting this error:

2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
/usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
Manager]Data source name not found, and no default driver specified

The line it's failing on is:


env = assert (luasql.odbc());

Thanks in advance for letting me know what I'm doing wrong.

Best,
Neil

On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

>
>
> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Hi Lloyd,
>>
>> Even after the change to the config file below, I can't get the package to
>> compile:
>>
>> [root at topaz luasql-2.1.1]# make
>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol'
>> can not be used when making a shared object; recompile with -fPIC
>> src/luasql.o: could not read symbols: Bad value
>> collect2: ld returned 1 exit status
>> make: *** [src/odbc.so] Error 1
>>
>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>
>
> Here's
> the script I use.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Mar 29 00:41:18 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 29 Mar 2010 09:41:18 +0200
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <20100328011108.1eb9d4fd@anubis.defcon1>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<20100327193909.6a0e60a8@anubis.defcon1>
	<1269716735.12574.20.camel@local.freepabx.com>
	<20100327202715.699c382a@anubis.defcon1>
	<1269733582.12574.24.camel@local.freepabx.com>
	<20100328011108.1eb9d4fd@anubis.defcon1>
Message-ID: <65d96fc81003290041h1486cfcdp9501e6690ef09f1a@mail.gmail.com>

hello,


thanks for your feedback... indeed, i need to encode the config file with
grandstream config tool and the phone will "understands" it :)

T.

On Sun, Mar 28, 2010 at 2:11 AM, Jean-Yves F. Barbier <12ukwn at gmail.com>wrote:

> Le Sat, 27 Mar 2010 23:46:22 +0000,
> David Knell <dave at 3c.co.uk> a ?crit :
>
> >
> > On Sat, 2010-03-27 at 20:27 +0100, Jean-Yves F. Barbier wrote:
> >
> > > At this time, the least critics I heard about IP phones was about SNOM
> > > (if some people could confirm that, it would be good:)
> >
> > Last had the pleasure of using one maybe two or three years ago - its
> > number auto-completion feature drove me nuts (dial 2002 to call someone
> > internally and, if you'd previously dialled 02072002123, it'd dial that
> > instead); I've no doubt you can turn it off, but I just returned to my
> > simple Grandstream.
>
> I see that this kind of appliance's the same as others: m$ malpractices,
> such as taking users for alpha-testers, has spread a lot :(
>
> --
> All men have the right to wait in line.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Mar 29 01:01:52 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 29 Mar 2010 10:01:52 +0200
Subject: [Freeswitch-users] grandstream gxp1200
In-Reply-To: <7d0bfd8c1003271559q5bca68b9p66efc11e389ee628@mail.gmail.com>
References: <65d96fc81003270150i13a09dcfg376e70179b8effb9@mail.gmail.com>
	<7d0bfd8c1003270310o4d407998o4a1beb5d1e8f9753@mail.gmail.com>
	<65d96fc81003271020o4b7f130bo65d39468ee296271@mail.gmail.com>
	<7d0bfd8c1003271559q5bca68b9p66efc11e389ee628@mail.gmail.com>
Message-ID: <65d96fc81003290101w164e24b8oe74a07c418806737@mail.gmail.com>

On Sun, Mar 28, 2010 at 12:59 AM, Nandy Dagondon <nandy1925 at gmail.com>wrote:

> > do i need to encode it in a certan way before putting it to a http
> server?
>
> do u mean, you just placed the edited configuration (text file) on the http
> server? no, you need to install the Configuration Tool program. the program
> needs the ff parameters:
> 1. MAC address
> 2. template text file
>
> it should produce a binary cfgxxxxxxxxxxxx file. this is the one u must
> place in your server.
>
> btw, i'm not familiar w/ DHCP Option 43. so, i use the keypad to point the
> IP phone to the config server. :-)
>


its done like this:

option vendor-encapsulated-options
42:19:68:74:74:70:3a:2f:2f:31:30:2e:34:2e:36:32:2e:37:2f:67:78:70:31:32:30:30:00;

This means http://10.4.62.7/gxp1200
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From jim at k4gvo.com  Sun Mar 28 14:59:37 2010
From: jim at k4gvo.com (Jim)
Date: Sun, 28 Mar 2010 17:59:37 -0400
Subject: [Freeswitch-users] How do I "register" a domain for a hard phone?
Message-ID: <4BAFD149.1060206@k4gvo.com>

I have just installed Freeswitch and have spent the day searching the 
wiki for an answer.  My soft phone on Linux and the Grand Stream hard 
phone are finally talking to the switch.  However the Grand Stream (on 
extension 101) requires the soft phone to call it as 101 at 192.168.2.99 
whereas the Grand Stream can call the softphone at extension 1000 
without an extension.  The soft phone is not on the same computer as the 
switch.  The configuration files are identical except for the extensions 
and passwords. 

I was getting a message like:
2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user 
[101 at 192.168.2.51]
You must define a domain called '192.168.2.51' in your directory and add 
a user with the id="101" attribute
and you must configure your device to use the proper domain in it's 
authentication credentials.

But I have no idea how to either correct this problem or set a domain 
anywhere.  There is no mention of domain in the configuration for the 
Grand Stream and changing the user to 101 at 192.168.2.51 in the phone and 
in the config didn't work.

Here are the respective config files:

<include>
  <user id="1000">
    <params>
      <param name="password" value="1234"/>
      <param name="vm-password" value="1234"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
      <variable name="outbound_caller_id_name" 
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" 
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

<include>
  <user id="101" mailbox="101">
    <params>
      <param name="password" value="secret"/>
      <param name="vm-password" value="1234"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="101"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 101"/>
      <variable name="effective_caller_id_number" value="101"/>
      <variable name="outbound_caller_id_name" 
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" 
value="$${outbound_caller_id}"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

Thanks for any assistance.

Jim.


From mike at jerris.com  Mon Mar 29 02:01:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Mar 2010 05:01:19 -0400
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <1269772725049-4812504.post@n2.nabble.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
Message-ID: <DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>

I'll say this one last thing, and then this thread is closed.  This same thread appears every few months on the mailing list, and its always the same content.  On multiple occasions people have setup one forum or another, no one reads them, no one responds on them, and they become completely useless, then a few months later, someone else comes around complaining how the forums are useless.  If someone is able and willing to do 100% of the work to add embeddable forums that link to our mailing lists and will require no additional work from us on an ongoing basis, feel free to contact me offlist with everything all setup, so I just need to login to the server and run a couple of commands to add it.

Mike


On Mar 28, 2010, at 6:38 AM, Ben Langfeld wrote:

> 
> "If you think there are shortcomings in the current Web interfaces then I'm
> sure it would be possible to add another on top of the existing mailing list
> setup."
> 
> Ok, so Nabble is a reasonable attempt but it still suffers from quoting
> issues and a failure to strip off all of the rubbish mailman footers on
> every post. It doesn't exactly scream minimalist and simple (DHH would have
> a fit) and that's my issue. For now, though, it will have to do.
> 
> Could we at least make a modification to the FreeSWITCH site to make people
> aware that these alternatives exist though? Having them hidden at crazy URLs
> isn't helping anyone. Nabble is supposedly embeddable - can we stick it in
> the main site and have it as prominent as the list?
> -- 
> View this message in context: http://n2.nabble.com/FreeSWITCH-Mailing-List-Why-not-a-forum-tp4808837p4812504.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.



From math.parent at gmail.com  Mon Mar 29 02:04:29 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Mon, 29 Mar 2010 11:04:29 +0200
Subject: [Freeswitch-users] mod_skinny
In-Reply-To: <87ACE9B862C84831AD6667AEFAD31F2C@bp1.ad.bp.com>
References: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com>
	<4BAF75E4.7050100@aktzero.com>
	<87ACE9B862C84831AD6667AEFAD31F2C@bp1.ad.bp.com>
Message-ID: <960738411003290204r7ea832b6ocb370ff471cdd1c@mail.gmail.com>

Hello,

> ----- Original Message -----
> From: Andrew Thompson
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sunday, March 28, 2010 4:29 PM
> Subject: Re: [Freeswitch-users] mod_skinny
> On 3/28/2010 11:08 AM, Dave Stevenson wrote:
>
> Hi,
>
> is mod_skinny (SCCP) still supported on FreeSWITCH ?
Of course! It is to young to be zombie ;)

I'm the dev of mod_skinny. The module works and the doc is probably
sufficient to setup a working installation (see
http://wiki.freeswitch.org/wiki/Mod_skinny).

But lot more is coming: I have patches to commit for:
- Shared lines
- multiple call per lines
- less crashes (I hope)

Next I will then work on:
- ForwardAll, ForwardBusy, ForwardNoAnswer
- Hold/Unhold
- Ring policies
- Transfer
- Proper memory pools and mutex releases

Some missing features that need help (or I don't care but know as important):
- Bypass Media
- Codec Negotiation
- answer VersionReqMessage with VersionMessage (easy)
- MusicOnHold

Feel free to contribute

More info on: http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#More

Also there are plenty of bugs, you can add them in JIRA.

On Sun, Mar 28, 2010 at 6:01 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Hi Andrew,
>
> thanks for the reply, under Windows, I don't have a \endpoints\mod_skinny
> directory though.

I have not tested on Windows if it even compiles, you probably need to
create a .vcproj (and submit it to JIRA).

It build on amd64 and ppc but probably behave incorrectly on ppc. You
can contribute on this even if you don't have a skinny phone: There is
a perl lib included to generate skinny messages (run it and send me
the capture).

If you plan to work on a feature, discuss on IRC or write it to
<http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>

Cheers

Mathieu Parent


From jason at jasonjgw.net  Mon Mar 29 02:06:58 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 29 Mar 2010 20:06:58 +1100
Subject: [Freeswitch-users] How do I "register" a domain for a hard
 phone?
In-Reply-To: <4BAFD149.1060206@k4gvo.com>
References: <4BAFD149.1060206@k4gvo.com>
Message-ID: <20100329090658.GA22955@jdc.jasonjgw.net>

Jim <jim at k4gvo.com> wrote:
 
> I was getting a message like:
> 2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user 
> [101 at 192.168.2.51]
> You must define a domain called '192.168.2.51' in your directory and add 
> a user with the id="101" attribute
> and you must configure your device to use the proper domain in it's 
> authentication credentials.
> 
> But I have no idea how to either correct this problem or set a domain 
> anywhere.  There is no mention of domain in the configuration for the 
> Grand Stream and changing the user to 101 at 192.168.2.51 in the phone and 
> in the config didn't work.

By default, FreeSWITCH uses its own IPv4 address as the domain. On my network,
for example, the address of the machine running FreeSWITCH is 192.168.0.2;
therefore, by default, it's looking for extension at 192.168.0.2, e.g.,
1001 at 192.168.0.2.

Suppose I want to configure my SIP phone to use extension 1001. To do this, I
enter 1001 in the user name field and 192.168.0.2 as the "registrar" in the
phone's configuration. The phone in this case is from Snom, but there will be
similar fields in the configurations of other hardware and software-based
phones.

I hope this is enough clarification that you can work out what to do in your
own particular scenario.



From mike at jerris.com  Mon Mar 29 02:42:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Mar 2010 05:42:07 -0400
Subject: [Freeswitch-users] Indian and mod_say
In-Reply-To: <8ccbff061003182132x697f6539jf7b5d6934c0924a9@mail.gmail.com>
References: <8ccbff061003182132x697f6539jf7b5d6934c0924a9@mail.gmail.com>
Message-ID: <627EAB3E-87FF-43E5-9DFB-B3163F14DC69@jerris.com>

translate docs dir xml file with all the phrases for recording, and conf for the phrase macros, write the c code for mod_say_in, get some prompts recorded (at least 48k masters) and then contact me offlist to discuss once all that is done and tested.

Mike

On Mar 19, 2010, at 12:32 AM, Dome Charoenyost wrote:

> Can someone guide me about Indian language? i want to create mod_say_in
> 



From hungngm at bkav.com.vn  Mon Mar 29 02:31:39 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Mon, 29 Mar 2010 16:31:39 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <A2F800DE8CD4F521DE9A65FDDBC775FF738F4800@hungngm>



Hi all.

I am newbegin and have 2 question as follow

1. Can config mod_fifo with member in external register ???

In my lab, i have OpenSIP as Proxy server and Register server and Freeswitch as
media server config mod_fifo (queue)

------------------- ---------------------

- opensips - - freeswitch -

8000 - - 8000 - -

---- ->- proxy server - ---------> - media server -

-register sv - - queue 8000 -

- - - -

-------------------- ---------------------

^

|

|

1010 at opensips_IP

User with extension 1010 is registerd with OpenSIPS server, queue 8000 is media
service in freeswitch server. Can i config mod_fifo in freeswitch with member in
Opensips server ?

I try config in /fifo.conf.xml like

<member timeout="15" simo="1"
lag="5">{fifo_member_wait=nowait}user/1010 at opensips_IP</member>

but it not work.

Someone done this with opensips or another register server, please post some
guide.

2. With mod_fifo: caller_id_name and caller_id_number will be replated by
default config:

origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'.
So, if caller_id_number is 0123456.. it will be replates wit 'fifo+test_fifo'. I
want origination_caller_id_number has value is 0123456..

Thank for help.

 

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From mike at jerris.com  Mon Mar 29 02:52:40 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 29 Mar 2010 05:52:40 -0400
Subject: [Freeswitch-users] Error Compiling FS With Enum Enabled
In-Reply-To: <3a2b3963e90f85624cdbecc95f854198.squirrel@www.thephinix.org>
References: <3a2b3963e90f85624cdbecc95f854198.squirrel@www.thephinix.org>
Message-ID: <BFE6DF00-09FC-4337-BA9B-22F11C368F52@jerris.com>

http://wiki.freeswitch.org/wiki/Reporting_Bugs

On Mar 24, 2010, at 7:21 AM, demuel at thephinix.org wrote:

> Hi,
> 
> I got this issue when compiling FS 1.0.5 under FreeBSD 7.0-Stable as show below:
> 
> _enum.c  -fPIC -DPIC -o .libs/mod_enum_la-mod_enum.o
> mod_enum.c: In function 'enum_lookup':
> mod_enum.c:457: error: 'dns_socket' undeclared (first use in this function)
> mod_enum.c:457: error: (Each undeclared identifier is reported only once
> mod_enum.c:457: error: for each function it appears in.)
> mod_enum.c:457: error: expected ';' before 'fd'
> cc1: warnings being treated as errors
> mod_enum.c:458: warning: ISO C90 forbids mixed declarations and code
> mod_enum.c:491: error: 'fd' undeclared (first use in this function)
> mod_enum.c: In function 'mod_enum_load':
> mod_enum.c:826: error: too few arguments to function 'dns_init'
> gmake[4]: *** [mod_enum_la-mod_enum.lo] Error 1
> gmake[3]: *** [mod_enum-all] Error 1
> gmake[2]: *** [all-recursive] Error 1
> 

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From maciej.aniserowicz at gmail.com  Mon Mar 29 03:01:35 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Mon, 29 Mar 2010 02:01:35 -0800 (PST)
Subject: [Freeswitch-users] Error when recording
Message-ID: <1269856895179-4817081.post@n2.nabble.com>


Hello,
I have a problem with recording sessions.

The scenario is:

1) dial 1st call and park
2) dial 2nd call and park
3) bridge them
4) start recording 2nd call
5) receive error message "Can not record session.  Media not enabled on
channel"

The log is here: http://pastebin.freeswitch.org/12566.

Both clients are FreeSWITCH extensions that play some wav file using
playback application defined in dialplan.
I get this error in ~25% of recorded sessions.

What am I doing wrong?

Thanks,
Maciej Aniserowicz
-- 
View this message in context: http://n2.nabble.com/Error-when-recording-tp4817081p4817081.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dujinfang at gmail.com  Mon Mar 29 03:14:04 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 29 Mar 2010 18:14:04 +0800
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <A2F800DE8CD4F521DE9A65FDDBC775FF738F4800@hungngm>
References: <A2F800DE8CD4F521DE9A65FDDBC775FF738F4800@hungngm>
Message-ID: <23f91031003290314k4ad509bcs77508d8c439d06a0@mail.gmail.com>

2010/3/29 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>:
> Hi all.
>
> I am newbegin and have 2 question as follow
>
> 1. Can config mod_fifo with member in external register ???
>
> ?In my lab, i have OpenSIP as Proxy server and Register server and
> Freeswitch as media server config mod_fifo (queue)
>
>
>
> ??????????? -------------------??????????????????????? ---------------------
>
> ??????????? - ? ?opensips?????????-??????????????????? -
> freeswitch????????????-
>
> 8000??? -?????????????????????? ???-??????? 8000??? -
> ??????? ?-
>
> ---- ->- proxy server????????- --------->?? - ?media server????????? -
>
> ??????????? -register sv ????????? -????????????????????- ?queue
> 8000??????????? -
>
> ??????????? -?????????????????????? ???-??????????????????? -
> ????????????????????? ?? ?????? -
>
> ??????????? --------------------????????????????????? ---------------------
>
> ??????????? ??????????? ^
>
> ??????????????????????? |
>
> ??????????????????????? |
>
> ??????????? ??1010 at opensips_IP
>
> User with extension 1010 is registerd with OpenSIPS server, queue 8000 is
> media service in freeswitch server. Can ?i config mod_fifo in freeswitch
> with member in Opensips server ?
>
> I try config in /fifo.conf.xml like
>
> <member timeout="15" simo="1"
> lag="5">{fifo_member_wait=nowait}user/1010 at opensips_IP</member>
>
> but it not work.
>

because it's not a valid dialstring. try

sofia/<your_profile_name>/ instead of user/

it's only a dialstring, so, if you can make "originate
<your_dial_string>" work then fifo should work.

> Someone done this with opensips or another register server, please post some
> guide.
>
>
>
> 2. With mod_fifo: caller_id_name and caller_id_number will be ?replated by
> default config:
>
> origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'.
> So, if caller_id_number is 0123456.. it will be replates wit
> 'fifo+test_fifo'. I want?origination_caller_id_number has value is
> 0123456..
>
no, because you  don't know which caller will actually been bridged to
which agent before the agent actually answer the call.


>
>
> Thank?for help.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

From lists at infosecurity.ch  Mon Mar 29 05:41:05 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 29 Mar 2010 14:41:05 +0200
Subject: [Freeswitch-users] FS auto nat detection systems: are an
 attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>
References: <4BAF250D.5010405@infosecurity.ch>
	<C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>
Message-ID: <4BB09FE1.8050003@infosecurity.ch>

Well,

i noticed that several way for FS to detect NAT does not detect NAT.

For example i have several operators that assign NATted ip address in 
network 1.0.0.0.0/8 that are not matched by apply-nat-acl=rfc1918.auto .
The only way to match it is to use aggressive-nat-detection.

But if you enable aggressive-nat-detection with PJSIP that have an 
auto-nat feature that detect an IP address change and re-register with 
adjusted Contact header, FS will not detected the host as NATted.

Other than this, i understood that FS have several dynamic methods to 
adjust the RTP relaying system.
While using VoIP under mobile environment, where latency could be very 
high, maybe some of this system just does not work because of some timer 
expires.

Now i am entering into deep analyzing of all those issues, however just 
now i can confirm that Asterisk with canreinvite=no handle several 
situation that FS with default-factory settings does not handle (when 
using mobile voip).

So for all the dynamic system of FS it would be really nice to be able 
to have some "static" method that just behave like the Asterisk 
canreinvite=no statically, without dynamic discovery methods where the 
heuristic could be easily tweaked/mismatch when running in particular 
environment such as mobile characterized by:
- High packet loss
- High Delay (especially in SIP signaling over TCP)
- NAT with translated address also in non-rfc1918 addresses

Just consider that to make working properly SIP over GPRS the T1 has to 
be setup to at least 3s instead of 500ms, that a full INVITE to be 
closed could take even 15-20s, so all the timing and timers that could 
exists around the FS dynamic system could break in such conditions 
(still not analyzed into the code).

Now i am going to arrange a test bed with Asterisk, FreeSWITCH and 
OpenSER+RTPProxy to see how different servers behave differently in 
various NAT situation with their stock factory default configuraiton 
and/or little changes.

Ciao,
Fabio Pietrosanti

On 28/03/10 12.03, Michael Jerris wrote:
> The situation you describe should not require anything not in the default configs of FreeSWITCH.  If its not working, please try reverting the configuration COMPLETELY back to the original configuration and it should work fine, then figure out what you changed to break it.
>
>
> On Mar 28, 2010, at 5:44 AM, Fabio Pietrosanti (naif) wrote:
>    
>> I am really struggling against the automatic nat system of FS, where the
>> full logic is still not really really clear and causing a lot of headheache.
>>
>>      
> No need to struggle, it works without changing anything.
>
>    
>> Where on asterisk with a simple canreinvite=no everything works fine
>> (even between dirty nat) on FS it does not work and FS start sending RTP
>> to provide ip address on 10.0.0.0/8 (even if there is
>> apply-nat-acl=rfc1918.auto) .
>>
>>      
> What configuration have you changed?
>    
Had to change NDLB-tls-connectile-dysfunction mandatory because other 
SIP/TLS does not work.
Without NDLB-tls-connectile-dysfunction FS try to open a socket to the 
NATted hosts that obviously cannot receive inbound TCP connections.

inbound-proxy-media=true
In order to simplify the handling of NAT i just enable this to force 
media relay and does not let bypass_media to be activated.

<param name="aggressive-nat-detection" value="true"/>
This is to detect conditions where the apply-nat-acl does not work 
because the Contact header reports private ip address in NAT where the 
IPs are not rfc1918 (2 mobile operator of 4 have this network setup).



>    
>> However, independently from the bugs of the automatic nat handling
>> systems that have with the time to be noticed, discovered, fixed, etc,
>> etc, my question is:
>>
>> All this method for dynamic nat detection and handling of FS, want to be
>> an alternative to the STUN/ICE/TURN approach of VoIP standard protocol?
>>      
> sometimes you simply can not avoid this approach, we have augmented it by throwing a little bit of upnp, natpmp, stun, and simple logic to get it right as much as it can.
>    

Probably will go also in that direction later but i think mainly to goes 
with RTP over TCP when UDP is blocked.

I think that, if we force the media relay, when outgoing UDP connection 
from the SIP UA are allowed everything should work right?

Fabio


From brian at freeswitch.org  Mon Mar 29 06:37:14 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 08:37:14 -0500
Subject: [Freeswitch-users] FS auto nat detection systems: are an
	attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <4BB09FE1.8050003@infosecurity.ch>
References: <4BAF250D.5010405@infosecurity.ch>
	<C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>
	<4BB09FE1.8050003@infosecurity.ch>
Message-ID: <5D2F352A-0EA9-45FD-A99D-5DD02EE53350@freeswitch.org>


On Mar 29, 2010, at 7:41 AM, Fabio Pietrosanti (naif) wrote:

> Well,
> 
> i noticed that several way for FS to detect NAT does not detect NAT.
> 
> For example i have several operators that assign NATted ip address in 
> network 1.0.0.0.0/8 that are not matched by apply-nat-acl=rfc1918.auto .
> The only way to match it is to use aggressive-nat-detection.

You shouldn't be using Aggressive-nat-detection.  If your operator is really using 1.0.0.0/8 then you need to add it to the rfc1918.auto space so it knows what is or is not nat.

> But if you enable aggressive-nat-detection with PJSIP that have an 
> auto-nat feature that detect an IP address change and re-register with 
> adjusted Contact header, FS will not detected the host as NATted.

Again see previous comment.

> Other than this, i understood that FS have several dynamic methods to 
> adjust the RTP relaying system.
> While using VoIP under mobile environment, where latency could be very 
> high, maybe some of this system just does not work because of some timer 
> expires.

What system do you speak of?  Auto Adj rtp?  Which has zip to do with a timer.  It has to do with X number of packets at the start of a call.

> Now i am entering into deep analyzing of all those issues, however just 
> now i can confirm that Asterisk with canreinvite=no handle several 
> situation that FS with default-factory settings does not handle (when 
> using mobile voip).

What?  We already keep the media flow thru FreeSWITCH which is the exact same behavior as canreinvite=no in Asterisk.

> So for all the dynamic system of FS it would be really nice to be able 
> to have some "static" method that just behave like the Asterisk 
> canreinvite=no statically, without dynamic discovery methods where the 
> heuristic could be easily tweaked/mismatch when running in particular 
> environment such as mobile characterized by:
> - High packet loss
> - High Delay (especially in SIP signaling over TCP)
> - NAT with translated address also in non-rfc1918 addresses

You clearly do not understand what canreinvite=no does.

> Just consider that to make working properly SIP over GPRS the T1 has to 
> be setup to at least 3s instead of 500ms, that a full INVITE to be 
> closed could take even 15-20s, so all the timing and timers that could 
> exists around the FS dynamic system could break in such conditions 
> (still not analyzed into the code).

Please read the code and fully understand it before making these wildly wrong statements.

> Now i am going to arrange a test bed with Asterisk, FreeSWITCH and 
> OpenSER+RTPProxy to see how different servers behave differently in 
> various NAT situation with their stock factory default configuraiton 
> and/or little changes.

FreeSWITCH will work in 90-95% of all NAT cases but thats going to depend on you knowing exactly how nat, sip and rtp works.  Without knowing these intricate details you'll fail.

/b



From brian at freeswitch.org  Mon Mar 29 06:40:07 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 08:40:07 -0500
Subject: [Freeswitch-users] need legA talks to listen leg B
In-Reply-To: <89203A70AD874529A0EA81B82D2C1E0A@MOBILEE1705>
References: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705>
	<4BAF0050.6000308@aktzero.com>
	<89203A70AD874529A0EA81B82D2C1E0A@MOBILEE1705>
Message-ID: <C0EBBAF6-DD2F-4C37-871E-1290B29693D1@freeswitch.org>


On Mar 28, 2010, at 3:03 AM, Madovsky wrote:

> with
> <action application="set" data="bridge_generate_comfort_noise=true"/> in profile

This isn't valid in the profile as you say.  Application is a dialplan specific 

> and
> <action application="set" data="bridge_generate_comfort_noise=true"/> in dialplan
> and
> {bridge_generate_comfort_noise=true} in the bridge everything is ok

You could set it as a global in vars.xml

> Thanks !
>  
> Franck

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From brian at freeswitch.org  Mon Mar 29 06:41:57 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 08:41:57 -0500
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <1269856895179-4817081.post@n2.nabble.com>
References: <1269856895179-4817081.post@n2.nabble.com>
Message-ID: <736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>

Are you doing anything with proxy media or bypass media?

/b

On Mar 29, 2010, at 5:01 AM, Maciej Aniserowicz wrote:

> 
> Hello,
> I have a problem with recording sessions.
> 
> The scenario is:
> 
> 1) dial 1st call and park
> 2) dial 2nd call and park
> 3) bridge them
> 4) start recording 2nd call
> 5) receive error message "Can not record session.  Media not enabled on
> channel"
> 
> The log is here: http://pastebin.freeswitch.org/12566.
> 
> Both clients are FreeSWITCH extensions that play some wav file using
> playback application defined in dialplan.
> I get this error in ~25% of recorded sessions.
> 
> What am I doing wrong?
> 
> Thanks,
> Maciej Aniserowicz

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From lists at infosecurity.ch  Mon Mar 29 07:01:31 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 29 Mar 2010 16:01:31 +0200
Subject: [Freeswitch-users] FS auto nat detection systems: are an
 attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <5D2F352A-0EA9-45FD-A99D-5DD02EE53350@freeswitch.org>
References: <4BAF250D.5010405@infosecurity.ch>	<C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>	<4BB09FE1.8050003@infosecurity.ch>
	<5D2F352A-0EA9-45FD-A99D-5DD02EE53350@freeswitch.org>
Message-ID: <4BB0B2BB.7030605@infosecurity.ch>

On 29/03/10 15.37, Brian West wrote:
> You shouldn't be using Aggressive-nat-detection.  If your operator is really using 1.0.0.0/8 then you need to add it to the rfc1918.auto space so it knows what is or is not nat.
>    

I need a system that is able to handle automatically that situation, i 
cannot continuously find out new fancy netblocks used by operator for 
nat. So the apply-nat-acl=rfc1918 without aggressive-nat-detection 
became not useful in detecting NAT.

>> But if you enable aggressive-nat-detection with PJSIP that have an
>> auto-nat feature that detect an IP address change and re-register with
>> adjusted Contact header, FS will not detected the host as NATted.
>>      
> Again see previous comment.
>    
Sure, but that's clearly an incompatibility of a client feature (that 
detect address change and re-register with proper contact header with 
the public ip address) with a server feature that make nat detection 
with apply-nat-acl and aggressive-nat-detection fail.

>> Other than this, i understood that FS have several dynamic methods to
>> adjust the RTP relaying system.
>> While using VoIP under mobile environment, where latency could be very
>> high, maybe some of this system just does not work because of some timer
>> expires.
>>      
> What system do you speak of?  Auto Adj rtp?  Which has zip to do with a timer.  It has to do with X number of packets at the start of a call.
>    

That's the part i don't know precisely. It's quite difficult to do RTP 
debugging on FS, i use tcpdump and the maximum debug level.
Auto adjust RTP i think it's the feature we are speaking about that 
require X packets within X time to start and need the SIP UA to start 
RTP as earlier as possible.
In a bandwidth saving environment like mobile you have to consume less 
bandwidth as possible starting the RTP flow only when SIP handshake has 
been completed, enable VAD, enable CNG and all sort of bandwidth 
optimization techniques.

It's possible to know precisely auto adjust rtp how it works precisely?
I have seen *really* strange scenarios with FS relaying RTP packets to 
provide IP addresses in situation when there was a very high packet loss 
and dropping of packets from the SIP UA with the private ip address.

>> Now i am entering into deep analyzing of all those issues, however just
>> now i can confirm that Asterisk with canreinvite=no handle several
>> situation that FS with default-factory settings does not handle (when
>> using mobile voip).
>>      
> What?  We already keep the media flow thru FreeSWITCH which is the exact same behavior as canreinvite=no in Asterisk.
>    
I tell you that's not true, are different software and it's reasonable 
that works differently.
Asterisk with canreinvite=no can handle certain mobile to mobile 
communication that FS with the minor modification i have done (also with 
the factory settings) does not handle.

Now i need carry on precise and structured testing to show off the 
different kind of servers how react to the different conditions and then 
analyze deeply how they are different.

I personally strongly prefer FS to Asterisk but have to say that 
Asterisk with static settings canreinvite=no is more reliable than FS 
with the dynamic detection system (NAT detection, RTP adjusting, etc) 
that are prone to fail in some extreme/particular condition.

> Please read the code and fully understand it before making these wildly wrong statements.
>    
Unfortunately i am a good tlc, security and it guys but at programming i 
read piece of code just to isolate issues.
For sure the NAT handling have situation that does not precisely 
identify NATted host.
Then maybe the auto Rtp adjust may require some improvement.
I've seen FS sending packets to 10.0.0.0 network even with an host that 
had AUTO-NAT in the "sofia status profile internal", so the RTP 
packetizer was sending data to the SIP UA internal address that was 
correctly identified as NATted looking at the Sofia information.

>> Now i am going to arrange a test bed with Asterisk, FreeSWITCH and
>> OpenSER+RTPProxy to see how different servers behave differently in
>> various NAT situation with their stock factory default configuraiton
>> and/or little changes.
>>      
> FreeSWITCH will work in 90-95% of all NAT cases but thats going to depend on you knowing exactly how nat, sip and rtp works.  Without knowing these intricate details you'll fail.
>    

I understand, it's an intricate situation, i think now i know quite 
well, SIP, RTP and network related issues.

For that reason i am trying to explain the particular context i 
encountered and i would like to arrange structured testing also to 
report to you guys the condition and details where it does not work 
properly and when on other server works.

It's all our common interests to make it working also in extreme 
condition :)

Fabio


From brian at freeswitch.org  Mon Mar 29 07:10:39 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 09:10:39 -0500
Subject: [Freeswitch-users] FS auto nat detection systems: are an
	attempt to provide alternative to STUN/ICE/TURN ?
In-Reply-To: <4BB0B2BB.7030605@infosecurity.ch>
References: <4BAF250D.5010405@infosecurity.ch>	<C14F1943-ECC1-4B36-8682-398345AF6C10@jerris.com>	<4BB09FE1.8050003@infosecurity.ch>
	<5D2F352A-0EA9-45FD-A99D-5DD02EE53350@freeswitch.org>
	<4BB0B2BB.7030605@infosecurity.ch>
Message-ID: <25237391-B8D7-45C5-B5C6-FB92BE58C811@freeswitch.org>

This is because your endpoint is an idiot and puts the wrong IP in the SDP... its going to cause us to send media to the wrong place until they send us a packet to auto adj.  It is a hard situation to solve and if your endpoint can't traverse nat on its own or figure out its info then you're out of luck.  Its not the registrars job or place to try to figure out your endpoints nat deficiency.  Your situation is a little out of the usual so I totally expect you to find all kinds of edge cases that aren't solved by default.  You do remember its SIP right?  The most retarded protocol on the planet. :P

/b

On Mar 29, 2010, at 9:01 AM, Fabio Pietrosanti (naif) wrote:

> Unfortunately i am a good tlc, security and it guys but at programming i 
> read piece of code just to isolate issues.
> For sure the NAT handling have situation that does not precisely 
> identify NATted host.
> Then maybe the auto Rtp adjust may require some improvement.
> I've seen FS sending packets to 10.0.0.0 network even with an host that 
> had AUTO-NAT in the "sofia status profile internal", so the RTP 
> packetizer was sending data to the SIP UA internal address that was 
> correctly identified as NATted looking at the Sofia information.



From brian at freeswitch.org  Mon Mar 29 07:18:51 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 09:18:51 -0500
Subject: [Freeswitch-users] ooh323 license is now compatible.
Message-ID: <99975897-3207-4FCF-AA61-4208FF46B9AB@freeswitch.org>

http://www.obj-sys.com/open/ooh323c_kits.php

The license of this library changed recently its compatible with ours now.  If someone wants to create an endpoint module using this h323 stack then have at it.

Just an FYI,

/b



From toofics at gmail.com  Mon Mar 29 07:21:53 2010
From: toofics at gmail.com (Victor Toofic)
Date: Mon, 29 Mar 2010 08:21:53 -0600
Subject: [Freeswitch-users] OneWay Bridge Channels
In-Reply-To: <191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
References: <1269632432.5486.54.camel@ktulu>
	<191c3a031003261531u3a3dd641h8dd1221a13e0d302@mail.gmail.com>
	<1269648734.5486.93.camel@ktulu>
	<191c3a031003261733q374c693es9b74e97136f623c6@mail.gmail.com>
Message-ID: <1269872513.5486.152.camel@ktulu>

It seems I've found a way to achieve this.

To pass audio from an answered channel B to an early-media channel A,
without answering A, you can do it as follows:

- When channel A arrives:
  - pre_answer
  - park
  - get its uuid

- Originate channel B:
  - get its uuid
  - uuid_transfer ChannelA-UUID eavesdrop:ChannelB-UUID inline

A little bit tricky, but it seems to work.

On Fri, 2010-03-26 at 19:33 -0500, Anthony Minessale wrote:
> no,
> 
> That sounds a little over-thought though?
> Why can't you just use the bridge app?  We have extensive parameters
> to do progress supervision on the wiki.
> 
> 
> On Fri, Mar 26, 2010 at 7:12 PM, Victor Toofic <toofics at gmail.com>
> wrote:
>         I don't get it. What is "<dialstring2>" for? I'm not
>         originating both
>         legs of the call.
>         
>         I already have the incoming A leg in the switch and I want it
>         to stay in
>         early media and "bridge" it one-way only to a originated B leg
>         that is
>         always answered since the very first second.
>         
>         What I am trying to do is some kind of call progress
>         supervision on the
>         B leg and answer and bridge to the A leg only when I detect
>         the B leg
>         was really answered. Before that I just want to pass the audio
>         coming
>         from the answered B leg to the early-media-state A leg.. so
>         the A leg
>         can hear the ringback tones, voice-mail messages, etc.
>         
>         I already detect the tones and when the B leg is really
>         answered, at
>         that moment I can do an uuid_bridge of both uuids and get the
>         legs
>         bridged. But I don't see how I could pass audio from B to A
>         before
>         answering the A leg.
>         
>         Is this possible in any way with FS??
>         
>         Maybe I didn't understand you, but I don't see how could it be
>         possible
>         to do it with an inline dialplan.
>         
>         Thnks!!
>         
>         
>         
>         On Fri, 2010-03-26 at 17:31 -0500, Anthony Minessale wrote:
>         > yes,
>         >
>         > don't park it, have it dial the other leg with the bridge
>         app.
>         >
>         > originate <dialstring1> bridge:<dialstring2> inline
>         >
>         >
>         > On Fri, Mar 26, 2010 at 2:40 PM, Victor Toofic
>         <toofics at gmail.com>
>         > wrote:
>         >         Hi!
>         >
>         >         Does anybody know if there is a way to "bridge" an
>         answered
>         >         channel to a
>         >         channel that is in "early media" state?
>         >
>         >         The scenario is like this:
>         >
>         >          Channel A               Channel B
>         >         -------------------> FS ------------->
>         >         (pre_answer, park)       (answer)
>         >         <---<---<---<---<--- Audio <---<---<---
>         >
>         >         When the channel A arrives I pre_answer and park it,
>         so it is
>         >         in "early
>         >         media" state. After that I originate the channel B.
>         When the
>         >         channel B
>         >         answers I just want to relay the audio coming from B
>         to A, I
>         >         don't care
>         >         if A can't talk to B at that moment.
>         >
>         >         Is there a way to achieve this? Some ideas?
>         >
>         >         Thnks!
>         >
>         >         -
>         >         Regards,
>         >         V?ctor Toofic
>         >
>         >
>         >
>         >         _______________________________________________
>         >         FreeSWITCH-users mailing list
>         >         FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org
>         >
>         >
>         >
>         > --
>         > Anthony Minessale II
>         >
>         > FreeSWITCH http://www.freeswitch.org/
>         > ClueCon http://www.cluecon.com/
>         > Twitter: http://twitter.com/FreeSWITCH_wire
>         >
>         > AIM: anthm
>         > MSN:anthony_minessale at hotmail.com
>         > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         > IRC: irc.freenode.net #freeswitch
>         >
>         > FreeSWITCH Developer Conference
>         > sip:888 at conference.freeswitch.org
>         > googletalk:conf+888 at conference.freeswitch.org
>         > pstn:+19193869900
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woodydickson at gmail.com  Mon Mar 29 07:25:34 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Mon, 29 Mar 2010 22:25:34 +0800
Subject: [Freeswitch-users] encrypted RTP
Message-ID: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>

Hi,

Is there anyway to transport RTP between FS and an endpoint in such a
way that the the FS server is going to defined a key and only the
endpoint that has the correct key can send and receive RTP from
Freeswitch?

Is there anyway to something that is similar?

thanks,
woody


From erkan at speedingtrade.com  Mon Mar 29 07:40:18 2010
From: erkan at speedingtrade.com (=?iso-8859-9?B?RXJrYW4g3G5s/A==?=)
Date: Mon, 29 Mar 2010 17:40:18 +0300
Subject: [Freeswitch-users] Set the outgoing caller id
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
	<87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>

The outgoing caller id don't work.

 

I try many different fields/variables and also analyze your web link about the Channel Variables.

The outgoing id works for internal calls.

But for external calls don't worked.

 

My provider support the outgoing caller id.

 

Here  my example from Asterisk. That works.

exten => _00XX.,1,Set(CALLERID(num)=00492222222)

exten => _00XX.,n,Set(CALLERID(name)=00492222222)

....

 

The variables caller_id_name  ande caller_id_number  are read only.

Can we open it so I can write into the fields, because with effective_caller_id_name and number don't work it.

Origination_caller...... are the same.

 

Hmmm... I don't understand where is my failure.

 

Kind regards

Erkan

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, March 26, 2010 5:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1"/>

Please try it and let us know what happens.
-MC

2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

Hi all,

 

I worked since 2 years with asterisk. But now I must use the Windows Platform.

So I found Freeswitch.

The first works with Freeswitch are very good, but I can't set the outgoing caller id.

 

The outgoing caller id is shown at the phone on the end of the line.

 

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

  <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

....

<include>

  <context name="default">

 

   <extension name="voicetrading.com">

     <condition field="destination_number" expression="^00(\d+)$">

         <action application="set" data="effective_caller_id_name=009012345678"/>

         <action application="set" data="effective_caller_id_number=009012345678"/>

         <action application="set" data="outbound_caller_id_name=009012345678"/>

         <action application="set" data="outbound_caller_id_number=009012345678"/>

         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

     </condition>

   </extension>

....

 

Can anyone give me a idea for what can I do.

 

Kind regards

Erkan Unlu


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From lloyd.aloysius at gmail.com  Mon Mar 29 07:55:40 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 29 Mar 2010 10:55:40 -0400
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>
	<e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com>
Message-ID: <8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com>

Did you uncomment from the following line in config file.


T= odbc

Thanks
Lloyd

On Mon, Mar 29, 2010 at 3:30 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks all, got it to compile and seems like my odbc connection is working
> properly:
>
> [root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
> +---------------------------------------+
> | Connected!                            |
> |                                       |
> | sql-statement                         |
> | help [tablename]                      |
> | quit                                  |
> |                                       |
> +---------------------------------------+
> SQL>
>
> In my lua script I have
>
> require "luasql.odbc"
>
> But am getting this error:
>
> 2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
> /usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
> Manager]Data source name not found, and no default driver specified
>
> The line it's failing on is:
>
> env = assert (luasql.odbc());
>
> Thanks in advance for letting me know what I'm doing wrong.
>
> Best,
> Neil
>
> On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>>
>>
>> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Hi Lloyd,
>>>
>>> Even after the change to the config file below, I can't get the package
>>> to compile:
>>>
>>> [root at topaz luasql-2.1.1]# make
>>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local
>>> symbol' can not be used when making a shared object; recompile with -fPIC
>>> src/luasql.o: could not read symbols: Bad value
>>> collect2: ld returned 1 exit status
>>> make: *** [src/odbc.so] Error 1
>>>
>>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>>
>>
>> Here's
>> the script I use.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Mon Mar 29 07:56:36 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 29 Mar 2010 16:56:36 +0200
Subject: [Freeswitch-users] Set the outgoing caller id
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
	<87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
Message-ID: <B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com>

Erkan,

have you tried to set:

<param name="caller-id-in-from" value="true"/>

in the gateway params ?

Also, your provider is supposed to tell you which field they will  
trust as the CLID.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :

> The outgoing caller id don?t work.
>
> I try many different fields/variables and also analyze your web link  
> about the Channel Variables.
> The outgoing id works for internal calls.
> But for external calls don?t worked.
>
> My provider support the outgoing caller id.
>
> Here  my example from Asterisk. That works.
> exten => _00XX.,1,Set(CALLERID(num)=00492222222)
> exten => _00XX.,n,Set(CALLERID(name)=00492222222)
> ?.
>
> The variables caller_id_name  ande caller_id_number  are read only.
> Can we open it so I can write into the fields, because with  
> effective_caller_id_name and number don?t work it.
> Origination_caller?? are the same.
>
> Hmmm? I don?t understand where is my failure.
>
> Kind regards
> Erkan
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Friday, March 26, 2010 5:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Some providers have different ways of handling this. Using  
> "effective_caller_id_number" and name are the usual ways of doing  
> it. However some providers pick up the caller ID from a different  
> field. Try experimenting with this:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type
>
> Perhaps a bridge line like this would work:
>
>   <action application="bridge" data="{sip_cid_type=rpid}sofia/ 
> gateway/voicetrading.com/00000$1"/>
>
> OR
>
>   <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/ 
> voicetrading.com/00000$1"/>
>
> Please try it and let us know what happens.
> -MC
>
> 2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>
> Hi all,
>
> I worked since 2 years with asterisk. But now I must use the Windows  
> Platform.
> So I found Freeswitch.
> The first works with Freeswitch are very good, but I can?t set the  
> outgoing caller id.
>
> The outgoing caller id is shown at the phone on the end of the line.
>
> I try some different variables but without result. My voip provider  
> support to set the callerid, in asterisk it works.
>   <action application="bridge" data="sofia/gateway/voicetrading.com/ 
> 00000$1"/>
> ?.
> <include>
>   <context name="default">
>
>    <extension name="voicetrading.com">
>      <condition field="destination_number" expression="^00(\d+)$">
>          <action application="set"  
> data="effective_caller_id_name=009012345678"/>
>          <action application="set"  
> data="effective_caller_id_number=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_name=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_number=009012345678"/>
>          <action application="bridge" data="sofia/gateway/ 
> voicetrading.com/00000$1"/>
>      </condition>
>    </extension>
> ?.
>
> Can anyone give me a idea for what can I do.
>
> Kind regards
> Erkan Unlu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From testa at voicetechnology.com.br  Mon Mar 29 08:04:15 2010
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 29 Mar 2010 12:04:15 -0300
Subject: [Freeswitch-users] Loopback echo test + conferencing
Message-ID: <9cb0e15e1003290804g77c5255xe0aee55669ec397d@mail.gmail.com>

Hi

I have a running conferencing application and need to measure the total
delay (FS+network).
I already have a way to measure delay using one FS instance, say FS1,
calling an echo application on FS2. This is achieved by recording the
session on FS1 and analyzing the cross correlation delay with a octave
script.
But how do I make FS1 call FS2 where FS2 has conference+echo so conference
is included on the test?

Thank you!

-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From anthony.minessale at gmail.com  Mon Mar 29 08:24:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Mar 2010 10:24:55 -0500
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
	<DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>
Message-ID: <191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>

I think the answer is in the fact that this thread generated 19+ replies.
which is more activity that the 2 existing forums have seen in the last 2
years.

It's always the same answer as Mike put it, and Andrew in that earlier
thread.  We need some help to do these things an d unless we do we have to
change as few things as possible to avoid overwhelming ourselves.

So everyone who is interested, show up Wednesday for the conference call at
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org> and
volunteer for some side projects.



On Mon, Mar 29, 2010 at 4:01 AM, Michael Jerris <mike at jerris.com> wrote:

> I'll say this one last thing, and then this thread is closed.  This same
> thread appears every few months on the mailing list, and its always the same
> content.  On multiple occasions people have setup one forum or another, no
> one reads them, no one responds on them, and they become completely useless,
> then a few months later, someone else comes around complaining how the
> forums are useless.  If someone is able and willing to do 100% of the work
> to add embeddable forums that link to our mailing lists and will require no
> additional work from us on an ongoing basis, feel free to contact me offlist
> with everything all setup, so I just need to login to the server and run a
> couple of commands to add it.
>
> Mike
>
>
> On Mar 28, 2010, at 6:38 AM, Ben Langfeld wrote:
>
> >
> > "If you think there are shortcomings in the current Web interfaces then
> I'm
> > sure it would be possible to add another on top of the existing mailing
> list
> > setup."
> >
> > Ok, so Nabble is a reasonable attempt but it still suffers from quoting
> > issues and a failure to strip off all of the rubbish mailman footers on
> > every post. It doesn't exactly scream minimalist and simple (DHH would
> have
> > a fit) and that's my issue. For now, though, it will have to do.
> >
> > Could we at least make a modification to the FreeSWITCH site to make
> people
> > aware that these alternatives exist though? Having them hidden at crazy
> URLs
> > isn't helping anyone. Nabble is supposedly embeddable - can we stick it
> in
> > the main site and have it as prominent as the list?
> > --
> > View this message in context:
> http://n2.nabble.com/FreeSWITCH-Mailing-List-Why-not-a-forum-tp4808837p4812504.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From oseslija at gmail.com  Mon Mar 29 08:37:10 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 29 Mar 2010 17:37:10 +0200
Subject: [Freeswitch-users] encrypted RTP
In-Reply-To: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>
References: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>
Message-ID: <4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>

http://wiki.freeswitch.org/wiki/SRTP

Ognjen

On Mon, Mar 29, 2010 at 4:25 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> Is there anyway to transport RTP between FS and an endpoint in such a
> way that the the FS server is going to defined a key and only the
> endpoint that has the correct key can send and receive RTP from
> Freeswitch?
>
> Is there anyway to something that is similar?
>
> thanks,
> woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Mar 29 08:41:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 10:41:27 -0500
Subject: [Freeswitch-users] encrypted RTP
In-Reply-To: <4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>
References: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>
	<4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>
Message-ID: <3167A9DA-3851-4BF2-AD77-E269675F495D@freeswitch.org>

He wants preshared keys or something like it.

/b

On Mar 29, 2010, at 10:37 AM, Ognjen Seslija wrote:

> http://wiki.freeswitch.org/wiki/SRTP
> 
> Ognjen

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From dave at 3c.co.uk  Mon Mar 29 08:45:49 2010
From: dave at 3c.co.uk (David Knell)
Date: Mon, 29 Mar 2010 09:45:49 -0600
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
	<DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>
	<191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>
Message-ID: <1269877549.12574.116.camel@local.freepabx.com>

Has anyone thought about a newsgroup?!

--Dave

> I think the answer is in the fact that this thread generated 19+
> replies. which is more activity that the 2 existing forums have seen
> in the last 2 years.
> 
> It's always the same answer as Mike put it, and Andrew in that earlier
> thread.  We need some help to do these things an d unless we do we
> have to change as few things as possible to avoid overwhelming
> ourselves.
> 
> So everyone who is interested, show up Wednesday for the conference
> call at sip:888 at conference.freeswitch.org and volunteer for some side
> projects.
> 
> 
> 
> On Mon, Mar 29, 2010 at 4:01 AM, Michael Jerris <mike at jerris.com>
> wrote:
>         I'll say this one last thing, and then this thread is closed.
>          This same thread appears every few months on the mailing
>         list, and its always the same content.  On multiple occasions
>         people have setup one forum or another, no one reads them, no
>         one responds on them, and they become completely useless, then
>         a few months later, someone else comes around complaining how
>         the forums are useless.  If someone is able and willing to do
>         100% of the work to add embeddable forums that link to our
>         mailing lists and will require no additional work from us on
>         an ongoing basis, feel free to contact me offlist with
>         everything all setup, so I just need to login to the server
>         and run a couple of commands to add it.
>         
>         Mike
>         
>         
>         
>         On Mar 28, 2010, at 6:38 AM, Ben Langfeld wrote:
>         
>         >
>         > "If you think there are shortcomings in the current Web
>         interfaces then I'm
>         > sure it would be possible to add another on top of the
>         existing mailing list
>         > setup."
>         >
>         > Ok, so Nabble is a reasonable attempt but it still suffers
>         from quoting
>         > issues and a failure to strip off all of the rubbish mailman
>         footers on
>         > every post. It doesn't exactly scream minimalist and simple
>         (DHH would have
>         > a fit) and that's my issue. For now, though, it will have to
>         do.
>         >
>         > Could we at least make a modification to the FreeSWITCH site
>         to make people
>         > aware that these alternatives exist though? Having them
>         hidden at crazy URLs
>         > isn't helping anyone. Nabble is supposedly embeddable - can
>         we stick it in
>         > the main site and have it as prominent as the list?
>         > --
>         > View this message in context:
>         http://n2.nabble.com/FreeSWITCH-Mailing-List-Why-not-a-forum-tp4808837p4812504.html
>         > Sent from the freeswitch-users mailing list archive at
>         Nabble.com.
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From neilp at cs.stanford.edu  Mon Mar 29 08:46:46 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 29 Mar 2010 08:46:46 -0700
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com> 
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com> 
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com> 
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com> 
	<e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com> 
	<8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com>
Message-ID: <e5503bdb1003290846s5e021a54j4211368ab105c7bb@mail.gmail.com>

Yes! I followed the instructions
here<http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#Manual_Installation>for
setting up ODBC, including linking the .ini files in freeswitch/etc.
Based on the test below it seems that the driver is working properly. Is the
code below correct (i.e. 'require "luasql.odbc";' and 'env =
assert(luasql.odbc());')?

Thanks,
Neil

On Mon, Mar 29, 2010 at 7:55 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Did you uncomment from the following line in config file.
>
>
> T= odbc
>
> Thanks
> Lloyd
>
> On Mon, Mar 29, 2010 at 3:30 AM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Thanks all, got it to compile and seems like my odbc connection is working
>> properly:
>>
>> [root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
>> +---------------------------------------+
>> | Connected!                            |
>> |                                       |
>> | sql-statement                         |
>> | help [tablename]                      |
>> | quit                                  |
>> |                                       |
>> +---------------------------------------+
>> SQL>
>>
>> In my lua script I have
>>
>> require "luasql.odbc"
>>
>> But am getting this error:
>>
>> 2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
>> /usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
>> Manager]Data source name not found, and no default driver specified
>>
>> The line it's failing on is:
>>
>> env = assert (luasql.odbc());
>>
>> Thanks in advance for letting me know what I'm doing wrong.
>>
>> Best,
>> Neil
>>
>> On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>
>>>
>>>
>>> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>
>>>> Hi Lloyd,
>>>>
>>>> Even after the change to the config file below, I can't get the package
>>>> to compile:
>>>>
>>>> [root at topaz luasql-2.1.1]# make
>>>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>>>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>>>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>>>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local
>>>> symbol' can not be used when making a shared object; recompile with -fPIC
>>>> src/luasql.o: could not read symbols: Bad value
>>>> collect2: ld returned 1 exit status
>>>> make: *** [src/odbc.so] Error 1
>>>>
>>>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>>>
>>>
>>> Here's
>>> the script I use.
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oseslija at gmail.com  Mon Mar 29 08:53:13 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 29 Mar 2010 17:53:13 +0200
Subject: [Freeswitch-users] encrypted RTP
In-Reply-To: <3167A9DA-3851-4BF2-AD77-E269675F495D@freeswitch.org>
References: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>
	<4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>
	<3167A9DA-3851-4BF2-AD77-E269675F495D@freeswitch.org>
Message-ID: <4468a6771003290853v7310f478g42fb275e04951494@mail.gmail.com>

I suggest using a tunnel before VoIP then.

O.

On Mon, Mar 29, 2010 at 5:41 PM, Brian West <brian at freeswitch.org> wrote:

> He wants preshared keys or something like it.
>
> /b
>
> On Mar 29, 2010, at 10:37 AM, Ognjen Seslija wrote:
>
> http://wiki.freeswitch.org/wiki/SRTP
>
> Ognjen
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Mon Mar 29 08:57:53 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 29 Mar 2010 11:57:53 -0400
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003290846s5e021a54j4211368ab105c7bb@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>
	<e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com>
	<8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com>
	<e5503bdb1003290846s5e021a54j4211368ab105c7bb@mail.gmail.com>
Message-ID: <8a19bf2e1003290857r6d919741k6e4e3369a4524ace@mail.gmail.com>

The config is correct. But that is UnixODBC configuration. Then you need to
load the LUA odbc driver for lua code access the UnixODBC.

Did you load the luasql.odbc?

The code is correct.

Thanks
Lloyd



On Mon, Mar 29, 2010 at 11:46 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Yes! I followed the instructions here<http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#Manual_Installation>for setting up ODBC, including linking the .ini files in freeswitch/etc.
> Based on the test below it seems that the driver is working properly. Is the
> code below correct (i.e. 'require "luasql.odbc";' and 'env =
> assert(luasql.odbc());')?
>
> Thanks,
> Neil
>
>
> On Mon, Mar 29, 2010 at 7:55 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> Did you uncomment from the following line in config file.
>>
>>
>> T= odbc
>>
>> Thanks
>> Lloyd
>>
>> On Mon, Mar 29, 2010 at 3:30 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Thanks all, got it to compile and seems like my odbc connection is
>>> working properly:
>>>
>>> [root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
>>> +---------------------------------------+
>>> | Connected!                            |
>>> |                                       |
>>> | sql-statement                         |
>>> | help [tablename]                      |
>>> | quit                                  |
>>> |                                       |
>>> +---------------------------------------+
>>> SQL>
>>>
>>> In my lua script I have
>>>
>>> require "luasql.odbc"
>>>
>>> But am getting this error:
>>>
>>> 2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
>>> /usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
>>> Manager]Data source name not found, and no default driver specified
>>>
>>> The line it's failing on is:
>>>
>>> env = assert (luasql.odbc());
>>>
>>> Thanks in advance for letting me know what I'm doing wrong.
>>>
>>> Best,
>>> Neil
>>>
>>> On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <mayamatakeshi at gmail.com
>>> > wrote:
>>>
>>>>
>>>>
>>>> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>
>>>>> Hi Lloyd,
>>>>>
>>>>> Even after the change to the config file below, I can't get the package
>>>>> to compile:
>>>>>
>>>>> [root at topaz luasql-2.1.1]# make
>>>>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>>>>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>>>>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>>>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>>>>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local
>>>>> symbol' can not be used when making a shared object; recompile with -fPIC
>>>>> src/luasql.o: could not read symbols: Bad value
>>>>> collect2: ld returned 1 exit status
>>>>> make: *** [src/odbc.so] Error 1
>>>>>
>>>>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>>>>
>>>>
>>>> Here's
>>>> the script I use.
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From fs-list at communicatefreely.net  Mon Mar 29 09:02:55 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 29 Mar 2010 12:02:55 -0400
Subject: [Freeswitch-users] mod_local_stream path based on rate
Message-ID: <4BB0CF2F.7070700@communicatefreely.net>

Hello,

I'm using local_stream:// as our music on hold source.  I have both 8000 KHz and 16000 KHz files
encoded so that I can play them at the native rate.  We have some endpoints that can only do PCMU
and others that can do higher rates.

I want to set music on hold for a channel and have it automatically pick the appropriate stream
based on the channel's native rate, just like the voice prompts do.  I tried creating a stream for
each rate, like rock-8000 and rock-16000 and then set the channel's music on hold variable to
local_stream://rock-${read_rate} but the read rate hasn't been negotiated when this is set in the
dialplan.

Any suggestions?

-Tim


From anthony.minessale at gmail.com  Mon Mar 29 09:16:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Mar 2010 11:16:16 -0500
Subject: [Freeswitch-users] mod_local_stream path based on rate
In-Reply-To: <4BB0CF2F.7070700@communicatefreely.net>
References: <4BB0CF2F.7070700@communicatefreely.net>
Message-ID: <191c3a031003290916o3b524d95p51eb637255afec20@mail.gmail.com>

the default config works this way
it runs a stream on each rate called moh/8000 moh/16000 moh/32000 and
moh/48000
if you try to run stream "moh" it will pick the right one.


On Mon, Mar 29, 2010 at 11:02 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hello,
>
> I'm using local_stream:// as our music on hold source.  I have both 8000
> KHz and 16000 KHz files
> encoded so that I can play them at the native rate.  We have some endpoints
> that can only do PCMU
> and others that can do higher rates.
>
> I want to set music on hold for a channel and have it automatically pick
> the appropriate stream
> based on the channel's native rate, just like the voice prompts do.  I
> tried creating a stream for
> each rate, like rock-8000 and rock-16000 and then set the channel's music
> on hold variable to
> local_stream://rock-${read_rate} but the read rate hasn't been negotiated
> when this is set in the
> dialplan.
>
> Any suggestions?
>
> -Tim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From neilp at cs.stanford.edu  Mon Mar 29 09:17:16 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 29 Mar 2010 09:17:16 -0700
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <8a19bf2e1003290857r6d919741k6e4e3369a4524ace@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com> 
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com> 
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com> 
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com> 
	<e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com> 
	<8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com> 
	<e5503bdb1003290846s5e021a54j4211368ab105c7bb@mail.gmail.com> 
	<8a19bf2e1003290857r6d919741k6e4e3369a4524ace@mail.gmail.com>
Message-ID: <e5503bdb1003290917tb3fb243pa8a8ae6f3456d772@mail.gmail.com>

Do you mean did I have this in my code?

require "luasql.odbc";

Yes, that is there.

On Mon, Mar 29, 2010 at 8:57 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> The config is correct. But that is UnixODBC configuration. Then you need to
> load the LUA odbc driver for lua code access the UnixODBC.
>
> Did you load the luasql.odbc?
>
> The code is correct.
>
> Thanks
> Lloyd
>
>
>
> On Mon, Mar 29, 2010 at 11:46 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>
>> Yes! I followed the instructions here<http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#Manual_Installation>for setting up ODBC, including linking the .ini files in freeswitch/etc.
>> Based on the test below it seems that the driver is working properly. Is the
>> code below correct (i.e. 'require "luasql.odbc";' and 'env =
>> assert(luasql.odbc());')?
>>
>> Thanks,
>> Neil
>>
>>
>> On Mon, Mar 29, 2010 at 7:55 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com
>> > wrote:
>>
>>> Did you uncomment from the following line in config file.
>>>
>>>
>>> T= odbc
>>>
>>> Thanks
>>> Lloyd
>>>
>>> On Mon, Mar 29, 2010 at 3:30 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>
>>>> Thanks all, got it to compile and seems like my odbc connection is
>>>> working properly:
>>>>
>>>> [root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
>>>> +---------------------------------------+
>>>> | Connected!                            |
>>>> |                                       |
>>>> | sql-statement                         |
>>>> | help [tablename]                      |
>>>> | quit                                  |
>>>> |                                       |
>>>> +---------------------------------------+
>>>> SQL>
>>>>
>>>> In my lua script I have
>>>>
>>>> require "luasql.odbc"
>>>>
>>>> But am getting this error:
>>>>
>>>> 2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
>>>> /usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
>>>> Manager]Data source name not found, and no default driver specified
>>>>
>>>> The line it's failing on is:
>>>>
>>>> env = assert (luasql.odbc());
>>>>
>>>> Thanks in advance for letting me know what I'm doing wrong.
>>>>
>>>> Best,
>>>> Neil
>>>>
>>>> On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <
>>>> mayamatakeshi at gmail.com> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>>
>>>>>> Hi Lloyd,
>>>>>>
>>>>>> Even after the change to the config file below, I can't get the
>>>>>> package to compile:
>>>>>>
>>>>>> [root at topaz luasql-2.1.1]# make
>>>>>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>>>>>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>>>>>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>>>>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>>>>>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local
>>>>>> symbol' can not be used when making a shared object; recompile with -fPIC
>>>>>> src/luasql.o: could not read symbols: Bad value
>>>>>> collect2: ld returned 1 exit status
>>>>>> make: *** [src/odbc.so] Error 1
>>>>>>
>>>>>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>>>>>
>>>>>
>>>>> Here's
>>>>> the script I use.
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From erkan at speedingtrade.com  Mon Mar 29 09:41:17 2010
From: erkan at speedingtrade.com (=?utf-8?B?RXJrYW4gw5xubMO8?=)
Date: Mon, 29 Mar 2010 19:41:17 +0300
Subject: [Freeswitch-users] Set the outgoing caller id
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
	<B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C53E3@server1.st.local>

Yes I set it to true, but no change.

 

I found another interesting thing.

 

If I understand it right.

 

<action application="set" data="dialed_number=00491234567890"/>

If I set this variable. So I change the goal of dialing number.

 

But the variable are not set.

 

If I change it in the bridge line

 

<action application="bridge" data="sofia/gateway/voicetrading.com/000004955555555"/>

 

So doesn?t matter which number I call I call everytime the 004955555555.

But the set of dialed_number must be give the same result but don?t work.

 

Maybe there exist another flag in another xml file which I must set to true so I can modify the variables?

 

Kind regards

Erkan

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, March 29, 2010 5:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Erkan,

 

have you tried to set:

 

<param name="caller-id-in-from" value="true"/>

 

in the gateway params ?

 

Also, your provider is supposed to tell you which field they will trust as the CLID.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :





The outgoing caller id don?t work.

 

I try many different fields/variables and also analyze your web link about the Channel Variables.

The outgoing id works for internal calls.

But for external calls don?t worked.

 

My provider support the outgoing caller id.

 

Here  my example from Asterisk. That works.

exten => _00XX.,1,Set(CALLERID(num)=00492222222)

exten => _00XX.,n,Set(CALLERID(name)=00492222222)

?.

 

The variables caller_id_name  ande caller_id_number  are read only.

Can we open it so I can write into the fields, because with effective_caller_id_name and number don?t work it.

Origination_caller?? are the same.

 

Hmmm? I don?t understand where is my failure.

 

Kind regards

Erkan

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, March 26, 2010 5:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1"/>

Please try it and let us know what happens.
-MC

2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

Hi all,

 

I worked since 2 years with asterisk. But now I must use the Windows Platform.

So I found Freeswitch.

The first works with Freeswitch are very good, but I can?t set the outgoing caller id.

 

The outgoing caller id is shown at the phone on the end of the line.

 

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

  <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

?.

<include>

  <context name="default">

 

   <extension name="voicetrading.com">

     <condition field="destination_number" expression="^00(\d+)$">

         <action application="set" data="effective_caller_id_name=009012345678"/>

         <action application="set" data="effective_caller_id_number=009012345678"/>

         <action application="set" data="outbound_caller_id_name=009012345678"/>

         <action application="set" data="outbound_caller_id_number=009012345678"/>

         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

     </condition>

   </extension>

?.

 

Can anyone give me a idea for what can I do.

 

Kind regards

Erkan Unlu


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Mon Mar 29 09:07:56 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 29 Mar 2010 12:07:56 -0400
Subject: [Freeswitch-users] How do I "register" a domain for a hard
	phone?
References: <4BAFD149.1060206@k4gvo.com>
Message-ID: <5995D5ACE7F3476CA02DDF4365D7F4F7@MOBILEE1705>

look at sip_profiles/internal.xml

----- Original Message ----- 
From: "Jim" <jim at k4gvo.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, March 28, 2010 5:59 PM
Subject: [Freeswitch-users] How do I "register" a domain for a hard phone?


>I have just installed Freeswitch and have spent the day searching the 
> wiki for an answer.  My soft phone on Linux and the Grand Stream hard 
> phone are finally talking to the switch.  However the Grand Stream (on 
> extension 101) requires the soft phone to call it as 101 at 192.168.2.99 
> whereas the Grand Stream can call the softphone at extension 1000 
> without an extension.  The soft phone is not on the same computer as the 
> switch.  The configuration files are identical except for the extensions 
> and passwords. 
> 
> I was getting a message like:
> 2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user 
> [101 at 192.168.2.51]
> You must define a domain called '192.168.2.51' in your directory and add 
> a user with the id="101" attribute
> and you must configure your device to use the proper domain in it's 
> authentication credentials.
> 
> But I have no idea how to either correct this problem or set a domain 
> anywhere.  There is no mention of domain in the configuration for the 
> Grand Stream and changing the user to 101 at 192.168.2.51 in the phone and 
> in the config didn't work.
> 
> Here are the respective config files:
> 
> <include>
>  <user id="1000">
>    <params>
>      <param name="password" value="1234"/>
>      <param name="vm-password" value="1234"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1000"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1000"/>
>      <variable name="effective_caller_id_number" value="1000"/>
>      <variable name="outbound_caller_id_name" 
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number" 
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> <include>
>  <user id="101" mailbox="101">
>    <params>
>      <param name="password" value="secret"/>
>      <param name="vm-password" value="1234"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="101"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 101"/>
>      <variable name="effective_caller_id_number" value="101"/>
>      <variable name="outbound_caller_id_name" 
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number" 
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> Thanks for any assistance.
> 
> Jim.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From infos at madovsky.org  Mon Mar 29 09:27:26 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 29 Mar 2010 12:27:26 -0400
Subject: [Freeswitch-users] need legA talks to listen leg B
References: <4938DA59AEE348D3866AE76091F8211F@MOBILEE1705><4BAF0050.6000308@aktzero.com><89203A70AD874529A0EA81B82D2C1E0A@MOBILEE1705>
	<C0EBBAF6-DD2F-4C37-871E-1290B29693D1@freeswitch.org>
Message-ID: <B1D613C18682460B92B6F1DFA0203255@MOBILEE1705>

Done, thank you 
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 29, 2010 9:40 AM
  Subject: Re: [Freeswitch-users] need legA talks to listen leg B




  On Mar 28, 2010, at 3:03 AM, Madovsky wrote:


    with
    <action application="set" data="bridge_generate_comfort_noise=true"/> in profile


  This isn't valid in the profile as you say.  Application is a dialplan specific 


    and
    <action application="set" data="bridge_generate_comfort_noise=true"/> in dialplan
    and
    {bridge_generate_comfort_noise=true} in the bridge everything is ok


  You could set it as a global in vars.xml



    Thanks !

    Franck




------------------------------------------------------------------------------


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From lloyd.aloysius at gmail.com  Mon Mar 29 10:06:28 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Mon, 29 Mar 2010 13:06:28 -0400
Subject: [Freeswitch-users] luasql on CentOS 5.4
In-Reply-To: <e5503bdb1003290917tb3fb243pa8a8ae6f3456d772@mail.gmail.com>
References: <e5503bdb1003262016r4313afc3jc01d2d3d55f872c7@mail.gmail.com>
	<09418D9B-A3F1-41D3-8823-AA20F103D083@freeswitch.org>
	<8a19bf2e1003270932q6c8bf862yfb99a460a8a49ee4@mail.gmail.com>
	<e5503bdb1003282144ofce64aane7df005f3c65d67a@mail.gmail.com>
	<15b9404e1003282203q5ba33af1pd4fa9a823ee0384e@mail.gmail.com>
	<e5503bdb1003290030s1662cd45g232c8599f59538b8@mail.gmail.com>
	<8a19bf2e1003290755y223cff17vc8378b18a4c9fcd7@mail.gmail.com>
	<e5503bdb1003290846s5e021a54j4211368ab105c7bb@mail.gmail.com>
	<8a19bf2e1003290857r6d919741k6e4e3369a4524ace@mail.gmail.com>
	<e5503bdb1003290917tb3fb243pa8a8ae6f3456d772@mail.gmail.com>
Message-ID: <8a19bf2e1003291006m755884f8le466791188d7cd77@mail.gmail.com>

No that is not my question.

Did you successfully install luasql with odbc driver enabled in the config
file?


On Mon, Mar 29, 2010 at 12:17 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Do you mean did I have this in my code?
>
> require "luasql.odbc";
>
> Yes, that is there.
>
>
> On Mon, Mar 29, 2010 at 8:57 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:
>
>> The config is correct. But that is UnixODBC configuration. Then you need
>> to load the LUA odbc driver for lua code access the UnixODBC.
>>
>> Did you load the luasql.odbc?
>>
>> The code is correct.
>>
>> Thanks
>> Lloyd
>>
>>
>>
>> On Mon, Mar 29, 2010 at 11:46 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Yes! I followed the instructions here<http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#Manual_Installation>for setting up ODBC, including linking the .ini files in freeswitch/etc.
>>> Based on the test below it seems that the driver is working properly. Is the
>>> code below correct (i.e. 'require "luasql.odbc";' and 'env =
>>> assert(luasql.odbc());')?
>>>
>>> Thanks,
>>> Neil
>>>
>>>
>>> On Mon, Mar 29, 2010 at 7:55 AM, Aloysius Lloyd <
>>> lloyd.aloysius at gmail.com> wrote:
>>>
>>>> Did you uncomment from the following line in config file.
>>>>
>>>>
>>>> T= odbc
>>>>
>>>> Thanks
>>>> Lloyd
>>>>
>>>> On Mon, Mar 29, 2010 at 3:30 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>
>>>>> Thanks all, got it to compile and seems like my odbc connection is
>>>>> working properly:
>>>>>
>>>>> [root at topaz bin]# isql maxpowersoft_odbc <myUser> <myPwd>
>>>>> +---------------------------------------+
>>>>> | Connected!                            |
>>>>> |                                       |
>>>>> | sql-statement                         |
>>>>> | help [tablename]                      |
>>>>> | quit                                  |
>>>>> |                                       |
>>>>> +---------------------------------------+
>>>>> SQL>
>>>>>
>>>>> In my lua script I have
>>>>>
>>>>> require "luasql.odbc"
>>>>>
>>>>> But am getting this error:
>>>>>
>>>>> 2010-03-28 22:21:42.080859 [ERR] mod_lua.cpp:182
>>>>> /usr/local/freeswitch/scripts/AO/paths.lua:49: [unixODBC][Driver
>>>>> Manager]Data source name not found, and no default driver specified
>>>>>
>>>>> The line it's failing on is:
>>>>>
>>>>> env = assert (luasql.odbc());
>>>>>
>>>>> Thanks in advance for letting me know what I'm doing wrong.
>>>>>
>>>>> Best,
>>>>> Neil
>>>>>
>>>>> On Sun, Mar 28, 2010 at 10:03 PM, mayamatakeshi <
>>>>> mayamatakeshi at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Mon, Mar 29, 2010 at 1:44 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>>>
>>>>>>> Hi Lloyd,
>>>>>>>
>>>>>>> Even after the change to the config file below, I can't get the
>>>>>>> package to compile:
>>>>>>>
>>>>>>> [root at topaz luasql-2.1.1]# make
>>>>>>> export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -fPIC -Wall
>>>>>>> -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src
>>>>>>> -DUNIXODBC -I/usr/local/include -I/usr/local/include  -o src/odbc.so -shared
>>>>>>>  src/luasql.o src/ls_odbc.o  -L/usr/local/lib -lodbc
>>>>>>> /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local
>>>>>>> symbol' can not be used when making a shared object; recompile with -fPIC
>>>>>>> src/luasql.o: could not read symbols: Bad value
>>>>>>> collect2: ld returned 1 exit status
>>>>>>> make: *** [src/odbc.so] Error 1
>>>>>>>
>>>>>>> Anything I need to change or do other than add -fPIC to CFLAGS?
>>>>>>>
>>>>>>
>>>>>> Here's
>>>>>> the script I use.
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testa at voicetechnology.com.br  Mon Mar 29 10:09:13 2010
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 29 Mar 2010 14:09:13 -0300
Subject: [Freeswitch-users] Loopback echo test + conferencing
In-Reply-To: <9cb0e15e1003290804g77c5255xe0aee55669ec397d@mail.gmail.com>
References: <9cb0e15e1003290804g77c5255xe0aee55669ec397d@mail.gmail.com>
Message-ID: <9cb0e15e1003291009j794012d6r22b54721f4187143@mail.gmail.com>

Just found myself:
originate sofia/external/3000 at 192.168.0.40:5080 &echo()
where 192.168.0.40 is my own IP, and enable incoming calls for destination
3000 where conferencing is taking place.

On Mon, Mar 29, 2010 at 12:04 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi
>
> I have a running conferencing application and need to measure the total
> delay (FS+network).
> I already have a way to measure delay using one FS instance, say FS1,
> calling an echo application on FS2. This is achieved by recording the
> session on FS1 and analyzing the cross correlation delay with a octave
> script.
> But how do I make FS1 call FS2 where FS2 has conference+echo so conference
> is included on the test?
>
> Thank you!
>
> --
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
>


-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From erkan at speedingtrade.com  Mon Mar 29 10:16:12 2010
From: erkan at speedingtrade.com (=?utf-8?B?RXJrYW4gw5xubMO8?=)
Date: Mon, 29 Mar 2010 20:16:12 +0300
Subject: [Freeswitch-users] Set the outgoing caller id
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
	<B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>

For internal calls between sipphones that work.

 

<extension name="internal">

   <condition field="destination_number" expression="^(\d{4})$">

         <action application="set" data="effective_caller_id_name=+49123456789"/>

         <action application="set" data="effective_caller_id_number=+4912345678"/>

       <action application="bridge" data="USER/$1@$${domain}"/>

   </condition>

 </extension>

 

 

But for external calls doesn?t work.

So set actions are proceed, but not for external calls.

 

I try different and more variables.

 

Kind regards.

Erkan

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, March 29, 2010 5:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Erkan,

 

have you tried to set:

 

<param name="caller-id-in-from" value="true"/>

 

in the gateway params ?

 

Also, your provider is supposed to tell you which field they will trust as the CLID.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :





The outgoing caller id don?t work.

 

I try many different fields/variables and also analyze your web link about the Channel Variables.

The outgoing id works for internal calls.

But for external calls don?t worked.

 

My provider support the outgoing caller id.

 

Here  my example from Asterisk. That works.

exten => _00XX.,1,Set(CALLERID(num)=00492222222)

exten => _00XX.,n,Set(CALLERID(name)=00492222222)

?.

 

The variables caller_id_name  ande caller_id_number  are read only.

Can we open it so I can write into the fields, because with effective_caller_id_name and number don?t work it.

Origination_caller?? are the same.

 

Hmmm? I don?t understand where is my failure.

 

Kind regards

Erkan

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, March 26, 2010 5:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1"/>

Please try it and let us know what happens.
-MC

2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

Hi all,

 

I worked since 2 years with asterisk. But now I must use the Windows Platform.

So I found Freeswitch.

The first works with Freeswitch are very good, but I can?t set the outgoing caller id.

 

The outgoing caller id is shown at the phone on the end of the line.

 

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

  <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

?.

<include>

  <context name="default">

 

   <extension name="voicetrading.com">

     <condition field="destination_number" expression="^00(\d+)$">

         <action application="set" data="effective_caller_id_name=009012345678"/>

         <action application="set" data="effective_caller_id_number=009012345678"/>

         <action application="set" data="outbound_caller_id_name=009012345678"/>

         <action application="set" data="outbound_caller_id_number=009012345678"/>

         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

     </condition>

   </extension>

?.

 

Can anyone give me a idea for what can I do.

 

Kind regards

Erkan Unlu


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From peter.olsson at visionutveckling.se  Mon Mar 29 10:28:11 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 29 Mar 2010 19:28:11 +0200
Subject: [Freeswitch-users] Set the outgoing caller id
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4B8C@cooper>

Most providers also have some kind of screening on caller id numbers. They will only allow your own numbers, and maybe in a specific format.

/Peter



________________________________
Fr?n: Erkan ?nl? <erkan at speedingtrade.com>
Skickat: den 29 mars 2010 19:23
Till: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] Set the outgoing caller id

For internal calls between sipphones that work.

<extension name="internal">
   <condition field="destination_number" expression="^(\d{4})$">
         <action application="set" data="effective_caller_id_name=+49123456789"/>
         <action application="set" data="effective_caller_id_number=+4912345678"/>
       <action application="bridge" data="USER/$1@$${domain}"/>
   </condition>
 </extension>


But for external calls doesn?t work.
So set actions are proceed, but not for external calls.

I try different and more variables.

Kind regards.
Erkan


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, March 29, 2010 5:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

Erkan,

have you tried to set:

<param name="caller-id-in-from" value="true"/>

in the gateway params ?

Also, your provider is supposed to tell you which field they will trust as the CLID.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr<mailto:david.ponzone at ipeva.fr>
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr<BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/>

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :


The outgoing caller id don?t work.

I try many different fields/variables and also analyze your web link about the Channel Variables.
The outgoing id works for internal calls.
But for external calls don?t worked.

My provider support the outgoing caller id.

Here  my example from Asterisk. That works.
exten => _00XX.,1,Set(CALLERID(num)=00492222222)
exten => _00XX.,n,Set(CALLERID(name)=00492222222)
?.

The variables caller_id_name  ande caller_id_number  are read only.
Can we open it so I can write into the fields, because with effective_caller_id_name and number don?t work it.
Origination_caller?? are the same.

Hmmm? I don?t understand where is my failure.

Kind regards
Erkan

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, March 26, 2010 5:37 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Set the outgoing caller id

Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/>

Please try it and let us know what happens.
-MC
2010/3/25 Erkan ?nl? <erkan at speedingtrade.com<mailto:erkan at speedingtrade.com>>
Hi all,

I worked since 2 years with asterisk. But now I must use the Windows Platform.
So I found Freeswitch.
The first works with Freeswitch are very good, but I can?t set the outgoing caller id.

The outgoing caller id is shown at the phone on the end of the line.

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.
  <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/>
?.
<include>
  <context name="default">

   <extension name="voicetrading.com<http://voicetrading.com>">
     <condition field="destination_number" expression="^00(\d+)$">
         <action application="set" data="effective_caller_id_name=009012345678"/>
         <action application="set" data="effective_caller_id_number=009012345678"/>
         <action application="set" data="outbound_caller_id_name=009012345678"/>
         <action application="set" data="outbound_caller_id_number=009012345678"/>
         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1<http://voicetrading.com/00000$1>"/>
     </condition>
   </extension>
?.

Can anyone give me a idea for what can I do.

Kind regards
Erkan Unlu

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4bb0e21032931608113207!

From jim at k4gvo.com  Mon Mar 29 10:29:35 2010
From: jim at k4gvo.com (Jim)
Date: Mon, 29 Mar 2010 13:29:35 -0400
Subject: [Freeswitch-users] How do I "register" a domain for a hard
	phone?
In-Reply-To: <20100329090658.GA22955@jdc.jasonjgw.net>
References: <4BAFD149.1060206@k4gvo.com>
	<20100329090658.GA22955@jdc.jasonjgw.net>
Message-ID: <4BB0E37F.3040403@k4gvo.com>

Jason White wrote:
> Jim <jim at k4gvo.com> wrote:
>  
>   
>> I was getting a message like:
>> 2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user 
>> [101 at 192.168.2.51]
>> You must define a domain called '192.168.2.51' in your directory and add 
>> a user with the id="101" attribute
>> and you must configure your device to use the proper domain in it's 
>> authentication credentials.
>>
>> But I have no idea how to either correct this problem or set a domain 
>> anywhere.  There is no mention of domain in the configuration for the 
>> Grand Stream and changing the user to 101 at 192.168.2.51 in the phone and 
>> in the config didn't work.
>>     
>
> By default, FreeSWITCH uses its own IPv4 address as the domain. On my network,
> for example, the address of the machine running FreeSWITCH is 192.168.0.2;
> therefore, by default, it's looking for extension at 192.168.0.2, e.g.,
> 1001 at 192.168.0.2.
>
> Suppose I want to configure my SIP phone to use extension 1001. To do this, I
> enter 1001 in the user name field and 192.168.0.2 as the "registrar" in the
> phone's configuration. The phone in this case is from Snom, but there will be
> similar fields in the configurations of other hardware and software-based
> phones.
>
> I hope this is enough clarification that you can work out what to do in your
> own particular scenario.
>   
Thanks and that's how it's set up.  My FS is on 192.168.2.51 and the 
Grand Stream is connecting just fine, however when I attempt to dial 
extension 101 from a phone it doesn't work.  The only connection I can 
make is by dialing 101 at 192.168.2.99.  I am guessing I have to translate 
101 into 101 at 192.168.2.99 somewhere.  I can only dial 101 from a soft 
phone which makes it somewhat unusable in most cases.

Jim.


From stevendt at primrosebank.net  Mon Mar 29 10:30:18 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Mon, 29 Mar 2010 18:30:18 +0100
Subject: [Freeswitch-users] mod_skinny
References: <56DA321343004AA9B236F6907DCF8AD9@bp1.ad.bp.com><4BAF75E4.7050100@aktzero.com><87ACE9B862C84831AD6667AEFAD31F2C@bp1.ad.bp.com>
	<960738411003290204r7ea832b6ocb370ff471cdd1c@mail.gmail.com>
Message-ID: <CEC58733230149A6971E8F54CAAC2142@bp1.ad.bp.com>

Hi Matthieu,

thanks for the info.

I have just picked up a Cisco SCCP phone and wanted to try it with 
FreeSWITCH. I won't be able to contribute to the code, but I will try to get 
it working on Windows with Visual Studio Express. I will rebuild FreeSWITCH 
at the weekend (I just ran out of time yesterday) and let you know how it 
goes,

regards
Dave




----- Original Message ----- 
From: "Mathieu Parent" <math.parent at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 29, 2010 10:04 AM
Subject: Re: [Freeswitch-users] mod_skinny


> Hello,
>
>> ----- Original Message -----
>> From: Andrew Thompson
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Sunday, March 28, 2010 4:29 PM
>> Subject: Re: [Freeswitch-users] mod_skinny
>> On 3/28/2010 11:08 AM, Dave Stevenson wrote:
>>
>> Hi,
>>
>> is mod_skinny (SCCP) still supported on FreeSWITCH ?
> Of course! It is to young to be zombie ;)
>
> I'm the dev of mod_skinny. The module works and the doc is probably
> sufficient to setup a working installation (see
> http://wiki.freeswitch.org/wiki/Mod_skinny).
>
> But lot more is coming: I have patches to commit for:
> - Shared lines
> - multiple call per lines
> - less crashes (I hope)
>
> Next I will then work on:
> - ForwardAll, ForwardBusy, ForwardNoAnswer
> - Hold/Unhold
> - Ring policies
> - Transfer
> - Proper memory pools and mutex releases
>
> Some missing features that need help (or I don't care but know as 
> important):
> - Bypass Media
> - Codec Negotiation
> - answer VersionReqMessage with VersionMessage (easy)
> - MusicOnHold
>
> Feel free to contribute
>
> More info on: 
> http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#More
>
> Also there are plenty of bugs, you can add them in JIRA.
>
> On Sun, Mar 28, 2010 at 6:01 PM, Dave Stevenson
> <stevendt at primrosebank.net> wrote:
>> Hi Andrew,
>>
>> thanks for the reply, under Windows, I don't have a \endpoints\mod_skinny
>> directory though.
>
> I have not tested on Windows if it even compiles, you probably need to
> create a .vcproj (and submit it to JIRA).
>
> It build on amd64 and ppc but probably behave incorrectly on ppc. You
> can contribute on this even if you don't have a skinny phone: There is
> a perl lib included to generate skinny messages (run it and send me
> the capture).
>
> If you plan to work on a feature, discuss on IRC or write it to
> <http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>
>
> Cheers
>
> Mathieu Parent
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From 12ukwn at gmail.com  Mon Mar 29 10:39:34 2010
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 29 Mar 2010 19:39:34 +0200
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
	<DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>
	<191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>
Message-ID: <20100329193934.34823762@anubis.defcon1>

Le Mon, 29 Mar 2010 10:24:55 -0500,
Anthony Minessale <anthony.minessale at gmail.com> a ?crit :

May be 1.0.1 HOWTOs should do the trick: the wiki's very good
(examples, commented vars, ...) BUT it is difficult to have an
overall look at FS for a newbie (like me, even if I've got some
experience in asterisk.)

From my point of view (may be only mine...), FS looks tremendously
powerful, but also terribly hard to understand its basics: info's here
but to much atomized - so the outcome may only be a gathering of all these
useful info in an *orderly* fashion.

> I think the answer is in the fact that this thread generated 19+ replies.
> which is more activity that the 2 existing forums have seen in the last 2
> years.
> 
> It's always the same answer as Mike put it, and Andrew in that earlier
> thread.  We need some help to do these things an d unless we do we have to
> change as few things as possible to avoid overwhelming ourselves.
> 
> So everyone who is interested, show up Wednesday for the conference call
> at sip:888 at conference.freeswitch.org
> <sip%3A888 at conference.freeswitch.org> and volunteer for some side
> projects.

-- 
All men are mortal.  Socrates was mortal.  Therefore, all men are Socrates.
		-- Woody Allen


From infos at madovsky.org  Mon Mar 29 10:25:20 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 29 Mar 2010 13:25:20 -0400
Subject: [Freeswitch-users] Set the outgoing caller id
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local><B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>
Message-ID: <C96DF5987CB2446BBF6FFF10BEEC04C1@MOBILEE1705>

try origination_caller_id_name and origination_caller_id_number
  ----- Original Message ----- 
  From: Erkan ?nl? 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, March 29, 2010 1:16 PM
  Subject: Re: [Freeswitch-users] Set the outgoing caller id


  For internal calls between sipphones that work.

   

  <extension name="internal">

     <condition field="destination_number" expression="^(\d{4})$">

           <action application="set" data="effective_caller_id_name=+49123456789"/>

           <action application="set" data="effective_caller_id_number=+4912345678"/>

         <action application="bridge" data="USER/$1@$${domain}"/>

     </condition>

   </extension>

   

   

  But for external calls doesn?t work.

  So set actions are proceed, but not for external calls.

   

  I try different and more variables.

   

  Kind regards.

  Erkan

   

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
  Sent: Monday, March 29, 2010 5:57 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Set the outgoing caller id

   

  Erkan,

   

  have you tried to set:

   

  <param name="caller-id-in-from" value="true"/>

   

  in the gateway params ?

   

  Also, your provider is supposed to tell you which field they will trust as the CLID.

   

  David Ponzone  Direction Technique

  email: david.ponzone at ipeva.fr

  tel:      01 74 03 18 97

  gsm:   06 66 98 76 34

   

  Service Client IPeva

  tel:      0811 46 26 26

  www.ipeva.fr  -   www.ipeva-studio.com

   

  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

   

   

   

   

  Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :





  The outgoing caller id don?t work.

   

  I try many different fields/variables and also analyze your web link about the Channel Variables.

  The outgoing id works for internal calls.

  But for external calls don?t worked.

   

  My provider support the outgoing caller id.

   

  Here  my example from Asterisk. That works.

  exten => _00XX.,1,Set(CALLERID(num)=00492222222)

  exten => _00XX.,n,Set(CALLERID(name)=00492222222)

  ?.

   

  The variables caller_id_name  ande caller_id_number  are read only.

  Can we open it so I can write into the fields, because with effective_caller_id_name and number don?t work it.

  Origination_caller?? are the same.

   

  Hmmm? I don?t understand where is my failure.

   

  Kind regards

  Erkan

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
  Sent: Friday, March 26, 2010 5:37 PM
  To: freeswitch-users at lists.freeswitch.org
  Subject: Re: [Freeswitch-users] Set the outgoing caller id

   

  Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

  http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

  Perhaps a bridge line like this would work:

    <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1"/>

  OR

    <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1"/>

  Please try it and let us know what happens.
  -MC

  2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

  Hi all,

   

  I worked since 2 years with asterisk. But now I must use the Windows Platform.

  So I found Freeswitch.

  The first works with Freeswitch are very good, but I can?t set the outgoing caller id.

   

  The outgoing caller id is shown at the phone on the end of the line.

   

  I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

    <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

  ?.

  <include>

    <context name="default">

   

     <extension name="voicetrading.com">

       <condition field="destination_number" expression="^00(\d+)$">

           <action application="set" data="effective_caller_id_name=009012345678"/>

           <action application="set" data="effective_caller_id_number=009012345678"/>

           <action application="set" data="outbound_caller_id_name=009012345678"/>

           <action application="set" data="outbound_caller_id_number=009012345678"/>

           <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

       </condition>

     </extension>

  ?.

   

  Can anyone give me a idea for what can I do.

   

  Kind regards

  Erkan Unlu


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org

   

  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org

   



------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From lists at infosecurity.ch  Mon Mar 29 10:56:39 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 29 Mar 2010 19:56:39 +0200
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
Message-ID: <4BB0E9D7.7050507@infosecurity.ch>

Hi,

does anyhave had experience in using FreeSWITCH along with STUN, ICE and 
TURN with the:
- Client implementation of PJSIP (STUN, ICE, TURN)
- Server implementation of FS + TURNSERVER.org (that provide STUN/TURN)

As per today the PJSIP guys have tested a working setup with:
- OpenSER/OpenSIPS
- TURNSERVER.org
- RTPProxy

Does anyone had experience in making the setup with FS + TURNSERVER ?

At that point i am thinking that the only choice i have given by 
default-to-extreme-scenario is not to use anymore the NAT detection and 
relaying features of FS but roll back to STUN/ICE/TURN solution using FS 
only for the SIP, but i would like to avoid learning another server 
product now we are fine with FS.

Fabio


From jim at k4gvo.com  Mon Mar 29 10:59:43 2010
From: jim at k4gvo.com (Jim)
Date: Mon, 29 Mar 2010 13:59:43 -0400
Subject: [Freeswitch-users] How do I "register" a domain for a hard
	phone?
In-Reply-To: <20100329090658.GA22955@jdc.jasonjgw.net>
References: <4BAFD149.1060206@k4gvo.com>
	<20100329090658.GA22955@jdc.jasonjgw.net>
Message-ID: <4BB0EA8F.8050501@k4gvo.com>

Jason White wrote:
> Jim <jim at k4gvo.com> wrote:
>  
>   
>> I was getting a message like:
>> 2010-03-28 13:46:52.514487 [WARNING] sofia_reg.c:1872 Can't find user 
>> [101 at 192.168.2.51]
>> You must define a domain called '192.168.2.51' in your directory and add 
>> a user with the id="101" attribute
>> and you must configure your device to use the proper domain in it's 
>> authentication credentials.
>>
>> But I have no idea how to either correct this problem or set a domain 
>> anywhere.  There is no mention of domain in the configuration for the 
>> Grand Stream and changing the user to 101 at 192.168.2.51 in the phone and 
>> in the config didn't work.
>>     
>
> By default, FreeSWITCH uses its own IPv4 address as the domain. On my network,
> for example, the address of the machine running FreeSWITCH is 192.168.0.2;
> therefore, by default, it's looking for extension at 192.168.0.2, e.g.,
> 1001 at 192.168.0.2.
>
> Suppose I want to configure my SIP phone to use extension 1001. To do this, I
> enter 1001 in the user name field and 192.168.0.2 as the "registrar" in the
> phone's configuration. The phone in this case is from Snom, but there will be
> similar fields in the configurations of other hardware and software-based
> phones.
>
> I hope this is enough clarification that you can work out what to do in your
> own particular scenario.
>
>   
The key was that I was using extensions outside of the 1000-1020 which 
are already defined in the dial  plan.  Once I started using extensions 
in the 1000-1019 things worked a lot better.

Thanks,
Jim.



From david.ponzone at gmail.com  Mon Mar 29 11:04:40 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 29 Mar 2010 20:04:40 +0200
Subject: [Freeswitch-users] Set the outgoing caller id
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
	<B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>
Message-ID: <76FC873D-CC2E-468C-8E88-73A142C7130A@gmail.com>

Erkan,

be aware that a lot of variables are available, all can be changed,  
but sometimes, it won't change anything.

Some variables will affect leg B, and should be defined with export  
app, not with set.
Some can be defined with the set app because they are automatically  
exported.
And some won't change anything on leg B, even if set.

In your case, I suspect your provider expects the CLID in a particular  
field, or as someone else said, they forbid you to send a CLID  
different from the DIDs they assigned to you.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/03/2010 ? 19:16, Erkan ?nl? a ?crit :

> For internal calls between sipphones that work.
>
> <extension name="internal">
>    <condition field="destination_number" expression="^(\d{4})$">
>          <action application="set" data="effective_caller_id_name= 
> +49123456789"/>
>          <action application="set" data="effective_caller_id_number= 
> +4912345678"/>
>        <action application="bridge" data="USER/$1@$${domain}"/>
>    </condition>
>  </extension>
>
>
> But for external calls doesn?t work.
> So set actions are proceed, but not for external calls.
>
> I try different and more variables.
>
> Kind regards.
> Erkan
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of David Ponzone
> Sent: Monday, March 29, 2010 5:57 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Erkan,
>
> have you tried to set:
>
> <param name="caller-id-in-from" value="true"/>
>
> in the gateway params ?
>
> Also, your provider is supposed to tell you which field they will  
> trust as the CLID.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :
>
>
> The outgoing caller id don?t work.
>
> I try many different fields/variables and also analyze your web link  
> about the Channel Variables.
> The outgoing id works for internal calls.
> But for external calls don?t worked.
>
> My provider support the outgoing caller id.
>
> Here  my example from Asterisk. That works.
> exten => _00XX.,1,Set(CALLERID(num)=00492222222)
> exten => _00XX.,n,Set(CALLERID(name)=00492222222)
> ?.
>
> The variables caller_id_name  ande caller_id_number  are read only.
> Can we open it so I can write into the fields, because with  
> effective_caller_id_name and number don?t work it.
> Origination_caller?? are the same.
>
> Hmmm? I don?t understand where is my failure.
>
> Kind regards
> Erkan
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Friday, March 26, 2010 5:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Some providers have different ways of handling this. Using  
> "effective_caller_id_number" and name are the usual ways of doing  
> it. However some providers pick up the caller ID from a different  
> field. Try experimenting with this:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type
>
> Perhaps a bridge line like this would work:
>
>   <action application="bridge" data="{sip_cid_type=rpid}sofia/ 
> gateway/voicetrading.com/00000$1"/>
>
> OR
>
>   <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/ 
> voicetrading.com/00000$1"/>
>
> Please try it and let us know what happens.
> -MC
>
> 2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>
> Hi all,
>
> I worked since 2 years with asterisk. But now I must use the Windows  
> Platform.
> So I found Freeswitch.
> The first works with Freeswitch are very good, but I can?t set the  
> outgoing caller id.
>
> The outgoing caller id is shown at the phone on the end of the line.
>
> I try some different variables but without result. My voip provider  
> support to set the callerid, in asterisk it works.
>   <action application="bridge" data="sofia/gateway/voicetrading.com/ 
> 00000$1"/>
> ?.
> <include>
>   <context name="default">
>
>    <extension name="voicetrading.com">
>      <condition field="destination_number" expression="^00(\d+)$">
>          <action application="set"  
> data="effective_caller_id_name=009012345678"/>
>          <action application="set"  
> data="effective_caller_id_number=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_name=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_number=009012345678"/>
>          <action application="bridge" data="sofia/gateway/ 
> voicetrading.com/00000$1"/>
>      </condition>
>    </extension>
> ?.
>
> Can anyone give me a idea for what can I do.
>
> Kind regards
> Erkan Unlu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From siniypin at gmail.com  Mon Mar 29 11:19:32 2010
From: siniypin at gmail.com (RobertT)
Date: Mon, 29 Mar 2010 13:19:32 -0500
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
Message-ID: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>

Currently I made this kind of setup. Everything, including pjsip based
client implementation, is as you described.
I use FS in bypass_media mode in order to be able to utilize all NAT
traversing capabilities possible while using pjsip with reTurnServer.
So far so good for me. But still, I have very limited traffic for now.

Best regards, RobertT.
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From msc at freeswitch.org  Mon Mar 29 11:26:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Mar 2010 11:26:46 -0700
Subject: [Freeswitch-users] Set the outgoing caller id
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local>
	<87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local>
Message-ID: <87f2f3b91003291126q39b56212u7272e291ea3374d@mail.gmail.com>

Get a pcap of the working caller ID setup on your Asterisk box and then
another pcap of the non-working setup on your FS box. Find out which field
the provider is using. Evidently they are doing something really stupid
because all of the methods that have been enumerated on this list have been
tested here on planet earth. :)

Here's some tips in case you need to get started with packet captures. Also,
Wireshark is handy for analyzing pcaps from tcpdump:

http://wiki.freeswitch.org/wiki/Packet_Capture

-MC

2010/3/29 Erkan ?nl? <erkan at speedingtrade.com>

>  The outgoing caller id don?t work.
>
>
>
> I try many different fields/variables and also analyze your web link about
> the Channel Variables.
>
> The outgoing id works for internal calls.
>
> But for external calls don?t worked.
>
>
>
> My provider support the outgoing caller id.
>
>
>
> Here  my example from Asterisk. That works.
>
> exten => _00XX.,1,Set(CALLERID(num)=00492222222)
>
> exten => _00XX.,n,Set(CALLERID(name)=00492222222)
>
> ?.
>
>
>
> The variables caller_id_name  ande caller_id_number  are read only.
>
> Can we open it so I can write into the fields, because with
> effective_caller_id_name and number don?t work it.
>
> Origination_caller?? are the same.
>
>
>
> Hmmm? I don?t understand where is my failure.
>
>
>
> Kind regards
>
> Erkan
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, March 26, 2010 5:37 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Set the outgoing caller id
>
>
>
> Some providers have different ways of handling this. Using
> "effective_caller_id_number" and name are the usual ways of doing it.
> However some providers pick up the caller ID from a different field. Try
> experimenting with this:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type
>
> Perhaps a bridge line like this would work:
>
>   <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/
> voicetrading.com/00000$1"/>
>
> OR
>
>   <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/
> voicetrading.com/00000$1"/>
>
> Please try it and let us know what happens.
> -MC
>
> 2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>
>
> Hi all,
>
>
>
> I worked since 2 years with asterisk. But now I must use the Windows
> Platform.
>
> So I found Freeswitch.
>
> The first works with Freeswitch are very good, but I can?t set the outgoing
> caller id.
>
>
>
> The outgoing caller id is shown at the phone on the end of the line.
>
>
>
> I try some different variables but without result. My voip provider support
> to set the callerid, in asterisk it works.
>
>   <action application="bridge" data="sofia/gateway/
> voicetrading.com/00000$1"/>
>
> ?.
>
> <include>
>
>   <context name="default">
>
>
>
>    <extension name="voicetrading.com">
>
>      <condition field="destination_number" expression="^00(\d+)$">
>
>          <action application="set"
> data="effective_caller_id_name=009012345678"/>
>
>          <action application="set"
> data="effective_caller_id_number=009012345678"/>
>
>          <action application="set"
> data="outbound_caller_id_name=009012345678"/>
>
>          <action application="set"
> data="outbound_caller_id_number=009012345678"/>
>
>          <action application="bridge" data="sofia/gateway/
> voicetrading.com/00000$1"/>
>
>      </condition>
>
>    </extension>
>
> ?.
>
>
>
> Can anyone give me a idea for what can I do.
>
>
>
> Kind regards
>
> Erkan Unlu
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From fraserredmond at gmail.com  Mon Mar 29 11:34:53 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 29 Mar 2010 19:34:53 +0100
Subject: [Freeswitch-users] media_bug_answer_req not working for me
Message-ID: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com>

Hi,

Has anyone used  media_bug_answer_req=true? Is there some trick to using it?
My recordings are happening fine, except that the ringtone is included at
the start.

This is the relevant parts of the dialplan - I've stripped out a bunch of
other things, and just included the things that could be relevant.

<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="call_timeout=20"/>
<action application="set"
data="storageName=${dialed_extension}_${strftime(%Y%m%d_%H%M)}"/>
<action application="set" data="storagePath=/opt/files/${strftime(%Y%m)}"/>
<action application="set" data="call_timeout=20"/>
<action application="export" data="nolocal:execute_on_answer=javascript
BridgeConnected.js"/>
<action application="javascript" data="BridgeBefore.js"/>

<action application="set" data="*media_bug_answer_req*=true"/>
<action application="record_session"
data="${storagePath}/${storageName}.wav"/>
<action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>


I'm on v1.0.5 (rev16966) so I shouldn't be using the old variable
(RECORD_ANSWER_REQ )

Cheers,
Fraser
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From brian at freeswitch.org  Mon Mar 29 11:41:51 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 13:41:51 -0500
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com>
Message-ID: <2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>

Could be because the variable is RECORD_ANSWER_REQ

/b

On Mar 29, 2010, at 1:34 PM, Fraser Redmond wrote:

> Hi,
> 
> Has anyone used  media_bug_answer_req=true? Is there some trick to using it? My recordings are happening fine, except that the ringtone is included at the start.
> 
> This is the relevant parts of the dialplan - I've stripped out a bunch of other things, and just included the things that could be relevant.
> 
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="continue_on_fail=true"/>
> <action application="set" data="call_timeout=20"/>
> <action application="set" data="storageName=${dialed_extension}_${strftime(%Y%m%d_%H%M)}"/>
> <action application="set" data="storagePath=/opt/files/${strftime(%Y%m)}"/>
> <action application="set" data="call_timeout=20"/>
> <action application="export" data="nolocal:execute_on_answer=javascript BridgeConnected.js"/>
> <action application="javascript" data="BridgeBefore.js"/>
> 
> <action application="set" data="media_bug_answer_req=true"/>
> <action application="record_session" data="${storagePath}/${storageName}.wav"/>
> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
> 
> 
> I'm on v1.0.5 (rev16966) so I shouldn't be using the old variable (RECORD_ANSWER_REQ )
> 
> Cheers,
> Fraser
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Mar 29 11:42:54 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 13:42:54 -0500
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
In-Reply-To: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>
References: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>
Message-ID: <78EA0977-A74A-4348-BEFB-F6596A4D8101@freeswitch.org>

We can't use any of the stuff in pjsip, He has some really nice nat traversal libs but they aren't license compatible.

/b

On Mar 29, 2010, at 1:19 PM, RobertT wrote:

> Currently I made this kind of setup. Everything, including pjsip based client implementation, is as you described.
> I use FS in bypass_media mode in order to be able to utilize all NAT traversing capabilities possible while using pjsip with reTurnServer.
> So far so good for me. But still, I have very limited traffic for now.
>  
> Best regards, RobertT.



From fraserredmond at gmail.com  Mon Mar 29 11:51:56 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 29 Mar 2010 19:51:56 +0100
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com> 
	<2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
Message-ID: <c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com>

Thanks, wiki says it changed:
http://wiki.freeswitch.org/wiki/Variable_media_bug_answer_req

I did try the old RECORD_ANSWER_REQ too, just in case (and have tried it
again now, to be sure), and it doesn't work either.

Cheers,
Fraser



On Mon, Mar 29, 2010 at 7:41 PM, Brian West <brian at freeswitch.org> wrote:

> Could be because the variable is RECORD_ANSWER_REQ
>
> /b
>
> On Mar 29, 2010, at 1:34 PM, Fraser Redmond wrote:
>
> Hi,
>
> Has anyone used  media_bug_answer_req=true? Is there some trick to using
> it? My recordings are happening fine, except that the ringtone is included
> at the start.
>
>
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From dome at tel.co.th  Mon Mar 29 12:05:26 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 30 Mar 2010 02:05:26 +0700
Subject: [Freeswitch-users] Regular Expression for match ip address
Message-ID: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>

Dear All,
   I want to autorize incoming call to public context by IP by
     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
but some time i want to allow 1 Class C or some range of IP. How can do that ?

BG

Dome C.


From brian at freeswitch.org  Mon Mar 29 12:15:59 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 14:15:59 -0500
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
Message-ID: <4E53C2F3-B612-42A1-B3DE-E279FCF0F6D9@freeswitch.org>

<condition field="${acl(${network_addr} 192.168.1.0/24)}" expression="true"/>

/b

On Mar 29, 2010, at 2:05 PM, Dome Charoenyost wrote:

> <condition field="network_addr" expression="^192\.168\.1\.1$"/>



From william.suffill at gmail.com  Mon Mar 29 12:16:12 2010
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 29 Mar 2010 15:16:12 -0400
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
Message-ID: <6b65470d1003291216n3b609fc3of50436e1e0419caf@mail.gmail.com>

You could use the ACL support and check_acl.

http://wiki.freeswitch.org/wiki/Acl#check_acl

which would probably scale slightly better since u can create an ACL list of
all hosts u want to treat the same way and just do 1 check_acl condition.

-- W
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From david.ponzone at gmail.com  Mon Mar 29 12:21:06 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 29 Mar 2010 21:21:06 +0200
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
Message-ID: <C5C4B58F-F996-4C43-96AC-148E1F77F0C8@gmail.com>

I am not sure to understand the question.
You want to allow some /32, one /24 and some other subnets to access  
FS and reach the public dialplan, while forbidding everything else ?

You can do that using ACL.

In acl.conf.xml, add this between <network-lists> and </network-lists>:
     <list name="external" default="deny">
       <node type="allow" cidr="192.168.1.1/32"/>
       <node type="allow" cidr="X.X.X.X/24"/>
        ...
     </list>

Assuming your only reachable SIP profile is external, you have to add  
in external.xml:
     <param name="apply-inbound-acl" value="external"/>

Modify this to match your configuration, but you get the idea.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/03/2010 ? 21:05, Dome Charoenyost a ?crit :

> Dear All,
>   I want to autorize incoming call to public context by IP by
>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> but some time i want to allow 1 Class C or some range of IP. How can  
> do that ?
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Mar 29 12:21:36 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 14:21:36 -0500
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <6b65470d1003291216n3b609fc3of50436e1e0419caf@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<6b65470d1003291216n3b609fc3of50436e1e0419caf@mail.gmail.com>
Message-ID: <F4844CE7-6751-490B-9D7D-0E59E73DF487@freeswitch.org>

Yah forgot about that one.. I like my ${acl()} usage much better

/b

On Mar 29, 2010, at 2:16 PM, William Suffill wrote:

> You could use the ACL support and check_acl.
> 
> http://wiki.freeswitch.org/wiki/Acl#check_acl
> 
> which would probably scale slightly better since u can create an ACL list of all hosts u want to treat the same way and just do 1 check_acl condition.
> 
> -- W

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From msc at freeswitch.org  Mon Mar 29 12:22:47 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Mar 2010 12:22:47 -0700
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
Message-ID: <87f2f3b91003291222n4c0b36e1t90c63f4f1bf79c8b@mail.gmail.com>

On Mon, Mar 29, 2010 at 12:05 PM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>   I want to autorize incoming call to public context by IP by
>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> but some time i want to allow 1 Class C or some range of IP. How can do
> that ?
>
>
Curious: can you not do this with acl.conf.xml? Must it be done in the
dialplan, that is, maybe you need to route calls differently because they
are from certain IP addresses? I'm just trying to understand the scope of
the problem.

-MC
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From infos at madovsky.org  Mon Mar 29 12:16:56 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 29 Mar 2010 15:16:56 -0400
Subject: [Freeswitch-users] Regular Expression for match ip address
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
Message-ID: <9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>

you need to use ACL in autoload_configs/acl.conf.xml

----- Original Message ----- 
From: "Dome Charoenyost" <dome at tel.co.th>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 29, 2010 3:05 PM
Subject: [Freeswitch-users] Regular Expression for match ip address


> Dear All,
>   I want to autorize incoming call to public context by IP by
>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> but some time i want to allow 1 Class C or some range of IP. How can do 
> that ?
>
> BG
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Mon Mar 29 13:30:36 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Mar 2010 15:30:36 -0500
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com>
	<2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
	<c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com>
Message-ID: <191c3a031003291330w575c2dfdre1eaccecdd557c50@mail.gmail.com>

did you check your logs and make sure your channel is not actually already
answered before you called bridge?


On Mon, Mar 29, 2010 at 1:51 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Thanks, wiki says it changed:
> http://wiki.freeswitch.org/wiki/Variable_media_bug_answer_req
>
> I did try the old RECORD_ANSWER_REQ too, just in case (and have tried it
> again now, to be sure), and it doesn't work either.
>
> Cheers,
> Fraser
>
>
>
>
> On Mon, Mar 29, 2010 at 7:41 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Could be because the variable is RECORD_ANSWER_REQ
>>
>> /b
>>
>> On Mar 29, 2010, at 1:34 PM, Fraser Redmond wrote:
>>
>> Hi,
>>
>> Has anyone used  media_bug_answer_req=true? Is there some trick to using
>> it? My recordings are happening fine, except that the ringtone is included
>> at the start.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From lists at infosecurity.ch  Mon Mar 29 13:38:10 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Mon, 29 Mar 2010 22:38:10 +0200
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
In-Reply-To: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>
References: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>
Message-ID: <4BB10FB2.9040509@infosecurity.ch>

Did you implemented this because of NAT problem with PJSIP along with 
FreeSWITCH?

With the FS dialplan you can still have the sip_nat_detected variable 
that let you set dynamically bypass_media or proxy_media depending on 
the NAT status of the peer.
That in theory is a dynamic method, much more simpler, yet in theory 
effective like ICE/TURN/STUN .

So my question is, do you had some specific issue with PJSIP along with 
FreeSWITCH to decide to introduce a Turn server along with FS?

Fabio

On 29/03/10 20.19, RobertT wrote:
> Currently I made this kind of setup. Everything, including pjsip based 
> client implementation, is as you described.
> I use FS in bypass_media mode in order to be able to utilize all NAT 
> traversing capabilities possible while using pjsip with reTurnServer.
> So far so good for me. But still, I have very limited traffic for now.
> Best regards, RobertT.
>



From erkan at speedingtrade.com  Mon Mar 29 13:45:50 2010
From: erkan at speedingtrade.com (=?utf-8?B?RXJrYW4gw5xubMO8?=)
Date: Mon, 29 Mar 2010 23:45:50 +0300
Subject: [Freeswitch-users] Set the outgoing caller id SOLVED
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local><B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>
	<76FC873D-CC2E-468C-8E88-73A142C7130A@gmail.com>
Message-ID: <81C2CEF80046FB4F863A60D4347DD33A0C53E5@server1.st.local>

Ok,

 

The problem is now solved.

         <action application="set" data="effective_caller_id_name=+4912345678"/>

         <action application="set" data="effective_caller_id_number=+4912345678"/>

 

That works really, in the gateway settings I must set 

 

<param name="caller-id-in-from" value="true"/>   Thanks to David.

 

All information?s that you give to me are correctly.

The reason why caller id don?t shown its I used every time reloadxml command for reload my new parameters.

This evening I shutdown and start my FS again. Now it works.

 

So I understand I must learn detailed what do reloadxml really.

 

Thank you to all and good evening.

 

Kind regards

Erkan

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, March 29, 2010 9:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Erkan,

 

be aware that a lot of variables are available, all can be changed, but sometimes, it won't change anything.

 

Some variables will affect leg B, and should be defined with export app, not with set.

Some can be defined with the set app because they are automatically exported.

And some won't change anything on leg B, even if set.

 

In your case, I suspect your provider expects the CLID in a particular field, or as someone else said, they forbid you to send a CLID different from the DIDs they assigned to you.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 29/03/2010 ? 19:16, Erkan ?nl? a ?crit :





For internal calls between sipphones that work.

 

<extension name="internal">

   <condition field="destination_number" expression="^(\d{4})$">

         <action application="set" data="effective_caller_id_name=+49123456789"/>

         <action application="set" data="effective_caller_id_number=+4912345678"/>

       <action application="bridge" data="USER/$1@$${domain}"/>

   </condition>

 </extension>

 

 

But for external calls doesn?t work.

So set actions are proceed, but not for external calls.

 

I try different and more variables.

 

Kind regards.

Erkan

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Monday, March 29, 2010 5:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Erkan,

 

have you tried to set:

 

<param name="caller-id-in-from" value="true"/>

 

in the gateway params ?

 

Also, your provider is supposed to tell you which field they will trust as the CLID.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :






The outgoing caller id don?t work.

 

I try many different fields/variables and also analyze your web link about the Channel Variables.

The outgoing id works for internal calls.

But for external calls don?t worked.

 

My provider support the outgoing caller id.

 

Here  my example from Asterisk. That works.

exten => _00XX.,1,Set(CALLERID(num)=00492222222)

exten => _00XX.,n,Set(CALLERID(name)=00492222222)

?.

 

The variables caller_id_name  ande caller_id_number  are read only.

Can we open it so I can write into the fields, because with effective_caller_id_name and number don?t work it.

Origination_caller?? are the same.

 

Hmmm? I don?t understand where is my failure.

 

Kind regards

Erkan

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, March 26, 2010 5:37 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Set the outgoing caller id

 

Some providers have different ways of handling this. Using "effective_caller_id_number" and name are the usual ways of doing it. However some providers pick up the caller ID from a different field. Try experimenting with this:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type

Perhaps a bridge line like this would work:

  <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/voicetrading.com/00000$1"/>

OR

  <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/voicetrading.com/00000$1"/>

Please try it and let us know what happens.
-MC

2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>

Hi all,

 

I worked since 2 years with asterisk. But now I must use the Windows Platform.

So I found Freeswitch.

The first works with Freeswitch are very good, but I can?t set the outgoing caller id.

 

The outgoing caller id is shown at the phone on the end of the line.

 

I try some different variables but without result. My voip provider support to set the callerid, in asterisk it works.

  <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

?.

<include>

  <context name="default">

 

   <extension name="voicetrading.com">

     <condition field="destination_number" expression="^00(\d+)$">

         <action application="set" data="effective_caller_id_name=009012345678"/>

         <action application="set" data="effective_caller_id_number=009012345678"/>

         <action application="set" data="outbound_caller_id_name=009012345678"/>

         <action application="set" data="outbound_caller_id_number=009012345678"/>

         <action application="bridge" data="sofia/gateway/voicetrading.com/00000$1"/>

     </condition>

   </extension>

?.

 

Can anyone give me a idea for what can I do.

 

Kind regards

Erkan Unlu


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon Mar 29 13:52:33 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 15:52:33 -0500
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
In-Reply-To: <4BB10FB2.9040509@infosecurity.ch>
References: <2160023e1003291119u4e0dea03s326d63bc42389dee@mail.gmail.com>
	<4BB10FB2.9040509@infosecurity.ch>
Message-ID: <579195B3-F716-49C8-841C-8BF50329F5DE@freeswitch.org>

If only PJNATH was licensed compatible.

/b

On Mar 29, 2010, at 3:38 PM, Fabio Pietrosanti (naif) wrote:

> Did you implemented this because of NAT problem with PJSIP along with 
> FreeSWITCH?
> 
> With the FS dialplan you can still have the sip_nat_detected variable 
> that let you set dynamically bypass_media or proxy_media depending on 
> the NAT status of the peer.
> That in theory is a dynamic method, much more simpler, yet in theory 
> effective like ICE/TURN/STUN .
> 
> So my question is, do you had some specific issue with PJSIP along with 
> FreeSWITCH to decide to introduce a Turn server along with FS?
> 
> Fabio



From gbraad at spotsnel.nl  Mon Mar 29 13:58:44 2010
From: gbraad at spotsnel.nl (Gerard Braad)
Date: Mon, 29 Mar 2010 22:58:44 +0200
Subject: [Freeswitch-users] domain instead of ip address on placed call...
Message-ID: <e8f45b9e1003291358h52c7060dq73b043f3a5278118@mail.gmail.com>

Hello Guys,


Have an issue with calls. I am testing this different devices (nokia
n900, nokia e60, nokia n810, ekiga, sipura spa3000) and have similar
results between them.
When I place a call from a device registered as 1001 at host.com to
1002 at host.com, it will show up at the other end (1002) as
1001 at 12.34.56.78
Can the IP address be replaced by the domain FQDN of the callers domain/acount?

greets,


Gerard


From fraserredmond at gmail.com  Mon Mar 29 14:03:39 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 29 Mar 2010 22:03:39 +0100
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <191c3a031003291330w575c2dfdre1eaccecdd557c50@mail.gmail.com>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com> 
	<2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
	<c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com> 
	<191c3a031003291330w575c2dfdre1eaccecdd557c50@mail.gmail.com>
Message-ID: <c32ee1cf1003291403x624bc28dkc049c06543a9df36@mail.gmail.com>

I've shifted the  <action application="set"
data="*media_bug_answer_req*=true"/>
to be the first action in the dialplan, and it's still not working.

Before, the aleg was answered, then the variable set and record_session set,
then the bridge was performed.

Cheers,
Fraser



On Mon, Mar 29, 2010 at 9:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> did you check your logs and make sure your channel is not actually already
> answered before you called bridge?
>
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From brian at freeswitch.org  Mon Mar 29 14:12:20 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 16:12:20 -0500
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <c32ee1cf1003291403x624bc28dkc049c06543a9df36@mail.gmail.com>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com>
	<2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
	<c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com>
	<191c3a031003291330w575c2dfdre1eaccecdd557c50@mail.gmail.com>
	<c32ee1cf1003291403x624bc28dkc049c06543a9df36@mail.gmail.com>
Message-ID: <AF8DB8AD-6627-4080-82B2-BF17DAD67B2C@freeswitch.org>

I have just confirmed it does indeed work.  Care to pastebin your logs and your dialplan again?

/b

On Mar 29, 2010, at 4:03 PM, Fraser Redmond wrote:

> I've shifted the  <action application="set" data="media_bug_answer_req=true"/>  to be the first action in the dialplan, and it's still not working.
> 
> Before, the aleg was answered, then the variable set and record_session set, then the bridge was performed.
> 
> Cheers,
> Fraser

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From gbraad at spotsnel.nl  Mon Mar 29 15:07:27 2010
From: gbraad at spotsnel.nl (Gerard Braad)
Date: Tue, 30 Mar 2010 00:07:27 +0200
Subject: [Freeswitch-users] issue with placing calls between users (uses
	server ip instead of domain)
Message-ID: <e8f45b9e1003291507i67f83bcfxe9fa85a8c545272b@mail.gmail.com>

I am testing the following with Freeswitch and different devices
(nokia n900, nokia e60, ekiga) and have similar results between them.

On the Freeswitch server (1.0.4 in multi-tenant mode) I have several
user profiles for a domain, e.g. 1000, 1001 for host.com
The user are authenticated correctly and calls can be placede. When I
place a call from a device registered as

   1001 at host.com

to

   1002 at host.com

it will show up at the other end (1002) as

   1001 at 12.34.56.78

I would expect this call to show up as 1001 at host.com. The IP address
is the one of from the Freeswitch server. Because of this, the calls
are no correctly recognized by the address book on certain devices.
Can the he domain FQDN of the callers domain/acount be used, instead
of the IP address of the server in the SIP uri?

kind regards,


Gerard


From brian at freeswitch.org  Mon Mar 29 15:13:15 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 17:13:15 -0500
Subject: [Freeswitch-users] issue with placing calls between users (uses
	server ip instead of domain)
In-Reply-To: <e8f45b9e1003291507i67f83bcfxe9fa85a8c545272b@mail.gmail.com>
References: <e8f45b9e1003291507i67f83bcfxe9fa85a8c545272b@mail.gmail.com>
Message-ID: <19464CA8-9EF7-4EB2-8AFB-D23B471D1618@freeswitch.org>

Well you could set the sip_invite_domain but again it doesn't really matter :P

/b

On Mar 29, 2010, at 5:07 PM, Gerard Braad wrote:

> I am testing the following with Freeswitch and different devices
> (nokia n900, nokia e60, ekiga) and have similar results between them.
> 
> On the Freeswitch server (1.0.4 in multi-tenant mode) I have several
> user profiles for a domain, e.g. 1000, 1001 for host.com
> The user are authenticated correctly and calls can be placede. When I
> place a call from a device registered as
> 
>   1001 at host.com
> 
> to
> 
>   1002 at host.com
> 
> it will show up at the other end (1002) as
> 
>   1001 at 12.34.56.78
> 
> I would expect this call to show up as 1001 at host.com. The IP address
> is the one of from the Freeswitch server. Because of this, the calls
> are no correctly recognized by the address book on certain devices.
> Can the he domain FQDN of the callers domain/acount be used, instead
> of the IP address of the server in the SIP uri?
> 
> kind regards,
> 
> 
> Gerard

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From david.ponzone at gmail.com  Mon Mar 29 15:29:25 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 00:29:25 +0200
Subject: [Freeswitch-users] Set the outgoing caller id SOLVED
In-Reply-To: <81C2CEF80046FB4F863A60D4347DD33A0C53E5@server1.st.local>
References: <81C2CEF80046FB4F863A60D4347DD33A0C53DC@server1.st.local><87f2f3b91003260837r54205cc7g2bb9c3663492c3a0@mail.gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E2@server1.st.local><B4163C76-2FC2-4D64-8C67-36DBFB9206DB@gmail.com><81C2CEF80046FB4F863A60D4347DD33A0C53E4@server1.st.local>
	<76FC873D-CC2E-468C-8E88-73A142C7130A@gmail.com>
	<81C2CEF80046FB4F863A60D4347DD33A0C53E5@server1.st.local>
Message-ID: <95BB1D49-D05E-4A66-AA7C-9B3146D4EFAF@gmail.com>

Ah sure.
When you change a parameter in a gateway config, you must do:
sofia profile <name of profile> killgw <name of gateway>
sofia profile <name of profile> rescan reloadxml

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/03/2010 ? 22:45, Erkan ?nl? a ?crit :

> Ok,
>
> The problem is now solved.
>          <action application="set" data="effective_caller_id_name= 
> +4912345678"/>
>          <action application="set" data="effective_caller_id_number= 
> +4912345678"/>
>
> That works really, in the gateway settings I must set
>
> <param name="caller-id-in-from" value="true"/>   Thanks to David.
>
> All information?s that you give to me are correctly.
> The reason why caller id don?t shown its I used every time  
> reloadxml command for reload my new parameters.
> This evening I shutdown and start my FS again. Now it works.
>
> So I understand I must learn detailed what do reloadxml really.
>
> Thank you to all and good evening.
>
> Kind regards
> Erkan
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of David Ponzone
> Sent: Monday, March 29, 2010 9:05 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Erkan,
>
> be aware that a lot of variables are available, all can be changed,  
> but sometimes, it won't change anything.
>
> Some variables will affect leg B, and should be defined with export  
> app, not with set.
> Some can be defined with the set app because they are automatically  
> exported.
> And some won't change anything on leg B, even if set.
>
> In your case, I suspect your provider expects the CLID in a  
> particular field, or as someone else said, they forbid you to send a  
> CLID different from the DIDs they assigned to you.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 29/03/2010 ? 19:16, Erkan ?nl? a ?crit :
>
>
> For internal calls between sipphones that work.
>
> <extension name="internal">
>    <condition field="destination_number" expression="^(\d{4})$">
>          <action application="set" data="effective_caller_id_name= 
> +49123456789"/>
>          <action application="set" data="effective_caller_id_number= 
> +4912345678"/>
>        <action application="bridge" data="USER/$1@$${domain}"/>
>    </condition>
>  </extension>
>
>
> But for external calls doesn?t work.
> So set actions are proceed, but not for external calls.
>
> I try different and more variables.
>
> Kind regards.
> Erkan
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of David Ponzone
> Sent: Monday, March 29, 2010 5:57 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Erkan,
>
> have you tried to set:
>
> <param name="caller-id-in-from" value="true"/>
>
> in the gateway params ?
>
> Also, your provider is supposed to tell you which field they will  
> trust as the CLID.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 29/03/2010 ? 16:40, Erkan ?nl? a ?crit :
>
>
>
> The outgoing caller id don?t work.
>
> I try many different fields/variables and also analyze your web link  
> about the Channel Variables.
> The outgoing id works for internal calls.
> But for external calls don?t worked.
>
> My provider support the outgoing caller id.
>
> Here  my example from Asterisk. That works.
> exten => _00XX.,1,Set(CALLERID(num)=00492222222)
> exten => _00XX.,n,Set(CALLERID(name)=00492222222)
> ?.
>
> The variables caller_id_name  ande caller_id_number  are read only.
> Can we open it so I can write into the fields, because with  
> effective_caller_id_name and number don?t work it.
> Origination_caller?? are the same.
>
> Hmmm? I don?t understand where is my failure.
>
> Kind regards
> Erkan
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Friday, March 26, 2010 5:37 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Set the outgoing caller id
>
> Some providers have different ways of handling this. Using  
> "effective_caller_id_number" and name are the usual ways of doing  
> it. However some providers pick up the caller ID from a different  
> field. Try experimenting with this:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type
>
> Perhaps a bridge line like this would work:
>
>   <action application="bridge" data="{sip_cid_type=rpid}sofia/ 
> gateway/voicetrading.com/00000$1"/>
>
> OR
>
>   <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/ 
> voicetrading.com/00000$1"/>
>
> Please try it and let us know what happens.
> -MC
>
> 2010/3/25 Erkan ?nl? <erkan at speedingtrade.com>
> Hi all,
>
> I worked since 2 years with asterisk. But now I must use the Windows  
> Platform.
> So I found Freeswitch.
> The first works with Freeswitch are very good, but I can?t set the  
> outgoing caller id.
>
> The outgoing caller id is shown at the phone on the end of the line.
>
> I try some different variables but without result. My voip provider  
> support to set the callerid, in asterisk it works.
>   <action application="bridge" data="sofia/gateway/voicetrading.com/ 
> 00000$1"/>
> ?.
> <include>
>   <context name="default">
>
>    <extension name="voicetrading.com">
>      <condition field="destination_number" expression="^00(\d+)$">
>          <action application="set"  
> data="effective_caller_id_name=009012345678"/>
>          <action application="set"  
> data="effective_caller_id_number=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_name=009012345678"/>
>          <action application="set"  
> data="outbound_caller_id_number=009012345678"/>
>          <action application="bridge" data="sofia/gateway/ 
> voicetrading.com/00000$1"/>
>      </condition>
>    </extension>
> ?.
>
> Can anyone give me a idea for what can I do.
>
> Kind regards
> Erkan Unlu
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fraserredmond at gmail.com  Mon Mar 29 15:52:49 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Mon, 29 Mar 2010 23:52:49 +0100
Subject: [Freeswitch-users] media_bug_answer_req not working for me
In-Reply-To: <AF8DB8AD-6627-4080-82B2-BF17DAD67B2C@freeswitch.org>
References: <c32ee1cf1003291134v1d6f977eh5786be688a0c4080@mail.gmail.com> 
	<2115690C-2E75-4E4B-8C72-685CF09DC570@freeswitch.org>
	<c32ee1cf1003291151k5766ca49n723aa7336e9a1b3d@mail.gmail.com> 
	<191c3a031003291330w575c2dfdre1eaccecdd557c50@mail.gmail.com> 
	<c32ee1cf1003291403x624bc28dkc049c06543a9df36@mail.gmail.com> 
	<AF8DB8AD-6627-4080-82B2-BF17DAD67B2C@freeswitch.org>
Message-ID: <c32ee1cf1003291552i57510dbk87f1a1679758cec5@mail.gmail.com>

Thanks Brian, mostly I just wanted some reassurance that it worked for
anyone - couldn't find many people referring to it, with it being such a new
variable.

I stripped my dialplan down to a test version that I could pastebin, then
stripped it down further until it started working, then added code back in
until I'd narrowed it down to this line being the one that breaks it:

<action application="set" data="ringback=${us-ring}"/>


Any idea why that would break it? Can that be set somewhere else or some
other way?


Here's the pastebin's in case it helps anything:

Dialplan xml  http://pastebin.freeswitch.org/12583

Log Output  http://pastebin.freeswitch.org/12584

Cheers,
Fraser




On Mon, Mar 29, 2010 at 10:12 PM, Brian West <brian at freeswitch.org> wrote:

> I have just confirmed it does indeed work.  Care to pastebin your logs and
> your dialplan again?
>
> /b
>
> On Mar 29, 2010, at 4:03 PM, Fraser Redmond wrote:
>
> I've shifted the  <action application="set" data="*media_bug_answer_req*=true"/>
> to be the first action in the dialplan, and it's still not working.
>
> Before, the aleg was answered, then the variable set and record_session
> set, then the bridge was performed.
>
> Cheers,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Mon Mar 29 16:35:45 2010
From: jason at jasonjgw.net (Jason White)
Date: Tue, 30 Mar 2010 10:35:45 +1100
Subject: [Freeswitch-users] FreeSWITCH Mailing List - Why not a forum?
In-Reply-To: <1269877549.12574.116.camel@local.freepabx.com>
References: <fb4ab1e41003270438s3cdf56d5t9b081923c8d68898@mail.gmail.com>
	<fb4ab1e41003271838s554e3065o6f8db802b14f6f4e@mail.gmail.com>
	<20100328015842.GA11026@jdc.jasonjgw.net>
	<1269772725049-4812504.post@n2.nabble.com>
	<DB7E7128-9B5D-4EDA-ABEC-454EF521BC71@jerris.com>
	<191c3a031003290824x28ad7d49o146a875b67b33cde@mail.gmail.com>
	<1269877549.12574.116.camel@local.freepabx.com>
Message-ID: <20100329233545.GA9477@jdc.jasonjgw.net>

David Knell <dave at 3c.co.uk> wrote:
> Has anyone thought about a newsgroup?!

nntp://news.gmane.org/gmane.comp.telephony.freeswitch.user

Unfortunately it seems not to be working properly - the latest posts there are
from late 2009 as far as I can tell, whereas it's working fine for other
mailing lists that I frequently use. In fact, I often read and post from there
instead of subscribing to the lists.



From infos at madovsky.org  Mon Mar 29 16:22:00 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 29 Mar 2010 19:22:00 -0400
Subject: [Freeswitch-users] domain instead of ip address on placed
	call...
References: <e8f45b9e1003291358h52c7060dq73b043f3a5278118@mail.gmail.com>
Message-ID: <B0B20AF27AE54F109A50A0B014590BCA@MOBILEE1705>

If I remember there are some threads that already exist
about this problem....

----- Original Message ----- 
From: "Gerard Braad" <gbraad at spotsnel.nl>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, March 29, 2010 4:58 PM
Subject: [Freeswitch-users] domain instead of ip address on placed call...


> Hello Guys,
>
>
> Have an issue with calls. I am testing this different devices (nokia
> n900, nokia e60, nokia n810, ekiga, sipura spa3000) and have similar
> results between them.
> When I place a call from a device registered as 1001 at host.com to
> 1002 at host.com, it will show up at the other end (1002) as
> 1001 at 12.34.56.78
> Can the IP address be replaced by the domain FQDN of the callers 
> domain/acount?
>
> greets,
>
>
> Gerard
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From woodydickson at gmail.com  Mon Mar 29 17:12:46 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Tue, 30 Mar 2010 08:12:46 +0800
Subject: [Freeswitch-users] encrypted RTP
In-Reply-To: <4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>
References: <a44688a1003290725q38b931d1i2c8b23fdae4725a8@mail.gmail.com>
	<4468a6771003290837r10e11965xcfaf9d1313a45d3c@mail.gmail.com>
Message-ID: <a44688a1003291712tc468a01l8257f64395151030@mail.gmail.com>

Is there anyway to limit the endpoints to be able to exchange media or
SIP messages with a particular server using  SRTP?

On Mon, Mar 29, 2010 at 11:37 PM, Ognjen Seslija <oseslija at gmail.com> wrote:
> http://wiki.freeswitch.org/wiki/SRTP
>
> Ognjen
>
> On Mon, Mar 29, 2010 at 4:25 PM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>>
>> Hi,
>>
>> Is there anyway to transport RTP between FS and an endpoint in such a
>> way that the the FS server is going to defined a key and only the
>> endpoint that has the correct key can send and receive RTP from
>> Freeswitch?
>>
>> Is there anyway to something that is similar?
>>
>> thanks,
>> woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sean at obscuradigital.com  Mon Mar 29 18:49:47 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Mon, 29 Mar 2010 18:49:47 -0700
Subject: [Freeswitch-users] Bandwidth.com
Message-ID: <C7D6A6CB.772C%sean@obscuradigital.com>

Hello,

Just received my bandwidth.com provisioning email with my sip_trunks and DID
numbers.  However, part of the instructions say I can?t use a username and
password when connecting to the gateway.  I tried leaving the username and
password blank in the gateway sip profile but it keeps giving me an error
message, Error: username param is REQUIRED.  Anyone ever setup a gateway
with Bandwidth and how does one get around the username param requirement.

Thanks
Sean
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From brian at freeswitch.org  Mon Mar 29 18:58:13 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 20:58:13 -0500
Subject: [Freeswitch-users] Bandwidth.com
In-Reply-To: <C7D6A6CB.772C%sean@obscuradigital.com>
References: <C7D6A6CB.772C%sean@obscuradigital.com>
Message-ID: <84FC5D72-2D35-4501-AC2F-D7B1EDE9BBFB@freeswitch.org>

in reality you don't need one... but you can put nobody in both.

/b

On Mar 29, 2010, at 8:49 PM, Sean Holt wrote:

> Hello,
> 
> Just received my bandwidth.com provisioning email with my sip_trunks and DID numbers.  However, part of the instructions say I can?t use a username and password when connecting to the gateway.  I tried leaving the username and password blank in the gateway sip profile but it keeps giving me an error message, Error: username param is REQUIRED.  Anyone ever setup a gateway with Bandwidth and how does one get around the username param requirement.
> 
> Thanks
> Sean

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From jaybinks at gmail.com  Mon Mar 29 19:01:25 2010
From: jaybinks at gmail.com (jay binks)
Date: Tue, 30 Mar 2010 12:01:25 +1000
Subject: [Freeswitch-users] Bandwidth.com
In-Reply-To: <C7D6A6CB.772C%sean@obscuradigital.com>
References: <C7D6A6CB.772C%sean@obscuradigital.com>
Message-ID: <ec2ce17e1003291901q7a280748v7fcc57ded2d3466c@mail.gmail.com>

I believe you need to still specify the username and password... but
use a "dummy" string.
from memory, they just get ignored but doing this satisfies freeswitch.


On Tue, Mar 30, 2010 at 11:49 AM, Sean Holt <sean at obscuradigital.com> wrote:
> Hello,
>
> Just received my bandwidth.com provisioning email with my sip_trunks and DID
> numbers. ?However, part of the instructions say I can?t use a username and
> password when connecting to the gateway. ?I tried leaving the username and
> password blank in the gateway sip profile but it keeps giving me an error
> message, Error: username param is REQUIRED. ?Anyone ever setup a gateway
> with Bandwidth and how does one get around the username param requirement.
>
> Thanks
> Sean
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely

Jay


From bjbrashier at gmail.com  Mon Mar 29 17:26:17 2010
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Mon, 29 Mar 2010 18:26:17 -0600
Subject: [Freeswitch-users] Call Recording & bind_meta_app
Message-ID: <7bcfdd291003291726g4b12a6bau7b990b2fae4d17b2@mail.gmail.com>

I need to add the ability to record a call to an existing
infrastructure, and I'm intending to use FreeSwitch to do this. The
recording needs to happen upon a DTMF command, which I understand is
normally done via bind_meta_app. From what I can tell, bind_meta_app
requires that you use the * key in addition to some other key.
Unfortunately, the system is already doing something else with *, so
that's not a key I can use to get this to work.

My question, then: what is the best way to record a call using DTMFs
that do not include *?

- Should I have Freeswitch forward an appropriate DTMF to a
controlling application that starts recording?

- Should I try to modify the bind_meta_app code to use another digit
instead (say, #)?  If so, where is the right place to look for this?

As a separate, but related question, I also need to play a disclaimer
when the recording starts ("This call is now being recorded."). Is it
OK to use bind_meta_app to bind 2 apps to the same DTMF combo?

BB


From neilp at cs.stanford.edu  Mon Mar 29 20:21:40 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 29 Mar 2010 20:21:40 -0700
Subject: [Freeswitch-users] debugging no audio I/O
Message-ID: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com>

Hi all,

I'm stuck trying to figure out why my lua app does not output any audio or
detect any DTMF input. This app has been installed and running successfully
on another server. The lua code in question is:

read(absolute_path_to_wav, 0);

Logs show:
2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-29 20:12:15.596856 [DEBUG] switch_ivr_play_say.c:1444 done playing
file

DTMF input is not being recognized, nothing is showing up in the logs. App
is being accessed with SIP from a remote client. I'm not really sure how to
debug this, so any pointers are appreciated. This is FS from trunk.

Thanks,
Neil
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From brian at freeswitch.org  Mon Mar 29 20:28:53 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 22:28:53 -0500
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com>
Message-ID: <52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>

what is this code? 

/b

On Mar 29, 2010, at 10:21 PM, Neil Patel wrote:

> read(absolute_path_to_wav, 0);



From neilp at cs.stanford.edu  Mon Mar 29 21:19:14 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 29 Mar 2010 21:19:14 -0700
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
Message-ID: <e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com>

oops, sorry about that. Here's read:

function read(file, delay)
   if (digits == "") then
      logfile:write(sessid, "\t",
    session:getVariable("caller_id_number"), "\t",
    os.time(), "\t", "Prompt", "\t", file, "\n");
      digits = session:read(1, 1, file, delay, "#");
      if (digits ~= "") then
 logfile:write(sessid, "\t", session:getVariable("caller_id_number"), "\t",
os.time(), "\t", "dtmf", "\t", file, "\t", digits, "\n");
      end
   end
end

On Mon, Mar 29, 2010 at 8:28 PM, Brian West <brian at freeswitch.org> wrote:

> what is this code?
>
> /b
>
> On Mar 29, 2010, at 10:21 PM, Neil Patel wrote:
>
> > read(absolute_path_to_wav, 0);
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cucku.cucku at yahoo.com.vn  Mon Mar 29 21:33:26 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Tue, 30 Mar 2010 12:33:26 +0800 (SGT)
Subject: [Freeswitch-users] need help on call to gateway
Message-ID: <49209.32043.qm@web76205.mail.sg1.yahoo.com>

Hi all

i use freeswitch to register to sip server
the domain is not real: sip.yeah.com
the outbound proxy : 118.69.145.5

i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
</include>
??? <gateway name="sip.yeah.com">
??? ??? <param name="username" value="071234"/>
??? ??? <param name="from-user" value="071234"/>??? 
??? ??? <param name="proxy" value="sip.yeah.com"/>
??? ??? <param name="username" value="071234"/>
??? ??? <param name="password" value="1234"/>
??? ??? <param name="register-proxy" value="118.69.145.5"/>
??? ??? <param name="register" value="true"/>
??? ??? <param name="retry-seconds" value="10"/>
??? </gateway>
</include>

so i use xlite register to freeswitch with user 1000
i start freeswitch , freeswitch start ok,
freeswitch register to sip server successfull with username 071234
xlite register to freeswitch successfull

so i edit the public.xml
for incomming call from sip server, i will forward to extension 1000
??? <extension name="sip.yeah.com">
????? <condition field="destination_number" expression="071234">
??????? <action application="transfer" data="1000 XML default"/>
????? </condition>
??? </extension>

but when i use xlite make call out from freeswitch to sip server and get no routing

nta: received INVITE sip:071234 at 192.168.1.90;user=phone SIP/2.0 (CSeq 3621)
nta: INVITE (3621) going to a default leg
nua(0xb6e20f90): adding session usage
nta: sent 100 Trying for INVITE (3621)
nua(0xb6e20f90): event i_invite 100 Trying
nua(0xb6e20f90): call state changed: init -> received, received offer
nua(0xb6e20f90): event i_state 100 Trying
432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006]
446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default
455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 192.168.1.90 to enum[071234 at default]
566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting
566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION]
nua(0xb6e20f90): recv signal r_respond 404 Not Found

i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch

please help, 

Thank you




      H?y c?ng chat v?i b?n b? tr?n Yahoo! Pingbox m?i toanh! T?o m?t kh?ng gian ri?ng ?? chat tr?n blog l? chuy?n nh?. http://vn.messenger.yahoo.com/pingbox/
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From brian at freeswitch.org  Mon Mar 29 21:37:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 29 Mar 2010 23:37:55 -0500
Subject: [Freeswitch-users] need help on call to gateway
In-Reply-To: <49209.32043.qm@web76205.mail.sg1.yahoo.com>
References: <49209.32043.qm@web76205.mail.sg1.yahoo.com>
Message-ID: <5A16ECE3-D398-4EA8-82CD-C09E6C3EB897@freeswitch.org>

its looking in context default... not public.

/b

On Mar 29, 2010, at 11:33 PM, false wrote:

> Hi all
> 
> i use freeswitch to register to sip server
> the domain is not real: sip.yeah.com
> the outbound proxy : 118.69.145.5
> 
> i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
> </include>
>     <gateway name="sip.yeah.com">
>         <param name="username" value="071234"/>
>         <param name="from-user" value="071234"/>    
>         <param name="proxy" value="sip.yeah.com"/>
>         <param name="username" value="071234"/>
>         <param name="password" value="1234"/>
>         <param name="register-proxy" value="118.69.145.5"/>
>         <param name="register" value="true"/>
>         <param name="retry-seconds" value="10"/>
>     </gateway>
> </include>
> 
> so i use xlite register to freeswitch with user 1000
> i start freeswitch , freeswitch start ok,
> freeswitch register to sip server successfull with username 071234
> xlite register to freeswitch successfull
> 
> so i edit the public.xml
> for incomming call from sip server, i will forward to extension 1000
>     <extension name="sip.yeah.com">
>       <condition field="destination_number" expression="071234">
>         <action application="transfer" data="1000 XML default"/>
>       </condition>
>     </extension>
> 
> but when i use xlite make call out from freeswitch to sip server and get no routing
> 
> nta: received INVITE sip:071234 at 192.168.1.90;user=phone SIP/2.0 (CSeq 3621)
> nta: INVITE (3621) going to a default leg
> nua(0xb6e20f90): adding session usage
> nta: sent 100 Trying for INVITE (3621)
> nua(0xb6e20f90): event i_invite 100 Trying
> nua(0xb6e20f90): call state changed: init -> received, received offer
> nua(0xb6e20f90): event i_state 100 Trying
> 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.90 [ef1af4b1-4d07-4b8e-8520-d99363347006]
> 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in context default
> 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 192.168.1.90 to enum[071234 at default]
> 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting
> 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 192.168.1.90 [CS_ROUTING] [NO_ROUTE_DESTINATION]
> nua(0xb6e20f90): recv signal r_respond 404 Not Found
> 
> i am not clear about routing call in freeswitch, i did read the dialplan in wiki but still not clear, is there any info to easy understand routing in freeswitch
> 
> please help, 
> 
> Thank you
> 
> 
> L??t web nhanh h?n 
> Internet Explorer 8 t?i ?u h?a cho Yahoo!, t? ??ng kh?i ??ng 2 trang b?n th?ch m?i l?n m? tr?nh duy?t. T?i IE8 t?i ??y! (Mi?n ph?)_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Mon Mar 29 23:35:07 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 08:35:07 +0200
Subject: [Freeswitch-users] Bandwidth.com
In-Reply-To: <C7D6A6CB.772C%sean@obscuradigital.com>
References: <C7D6A6CB.772C%sean@obscuradigital.com>
Message-ID: <74246E0C-E4F1-41B3-8232-172938099F80@gmail.com>

Sean,

as th other already said, you can put any string as the username and  
password, but I think you need to add this in the gateway params:

<param name="register" value="false"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/03/2010 ? 03:49, Sean Holt a ?crit :

> Hello,
>
> Just received my bandwidth.com provisioning email with my sip_trunks  
> and DID numbers.  However, part of the instructions say I can?t use  
> a username and password when connecting to the gateway.  I tried  
> leaving the username and password blank in the gateway sip profile  
> but it keeps giving me an error message, Error: username param is  
> REQUIRED.  Anyone ever setup a gateway with Bandwidth and how does  
> one get around the username param requirement.
>
> Thanks
> Sean
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Mon Mar 29 23:48:25 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 08:48:25 +0200
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com>
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com>
Message-ID: <5FE33791-3F80-4E82-933E-01558B036298@gmail.com>

Neil,

Are you sure you receive the DTMFs ?
You should see the DTMF events decoded by FS on the console (/log 7),  
even without any app reading them.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.


Le 30/03/2010 ? 06:19, Neil Patel a ?crit :

> oops, sorry about that. Here's read:
>
> function read(file, delay)
>    if (digits == "") then
>       logfile:write(sessid, "\t",
> 		    session:getVariable("caller_id_number"), "\t",
> 		    os.time(), "\t", "Prompt", "\t", file, "\n");
>       digits = session:read(1, 1, file, delay, "#");
>       if (digits ~= "") then
> 	 logfile:write(sessid, "\t",  
> session:getVariable("caller_id_number"), "\t", os.time(), "\t",  
> "dtmf", "\t", file, "\t", digits, "\n");
>       end
>    end
> end
>
> On Mon, Mar 29, 2010 at 8:28 PM, Brian West <brian at freeswitch.org>  
> wrote:
> what is this code?
>
> /b
>
> On Mar 29, 2010, at 10:21 PM, Neil Patel wrote:
>
> > read(absolute_path_to_wav, 0);
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Mon Mar 29 23:57:54 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 08:57:54 +0200
Subject: [Freeswitch-users] need help on call to gateway
In-Reply-To: <49209.32043.qm@web76205.mail.sg1.yahoo.com>
References: <49209.32043.qm@web76205.mail.sg1.yahoo.com>
Message-ID: <8D3F6BF2-E63E-472D-B50F-1B271DB002E5@gmail.com>

False,

The logic in FS is that a call will go through the dialplan context  
attached to the SIP profile that the call is coming into.
So if you have the SIP Profile internal bound to 10.0.0.1:5080 for  
instance, and then you receive an INVITE on 10.0.0.1:5080, FS will  
parse the dialplan context configured in internal.xml (by default,  
public).

For registered users though, you can alter this behavior by specifying  
a context in the user variables:

       <variable name="user_context"  
value="your_own_context_for_this_user"/>

If you don't, the context configured in the SIP profile that the user  
is sending INVITEs to will be used.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 30/03/2010 ? 06:33, false a ?crit :

> Hi all
>
> i use freeswitch to register to sip server
> the domain is not real: sip.yeah.com
> the outbound proxy : 118.69.145.5
>
> i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/ 
> sip_profiles/external folder
> </include>
>     <gateway name="sip.yeah.com">
>         <param name="username" value="071234"/>
>         <param name="from-user" value="071234"/>
>         <param name="proxy" value="sip.yeah.com"/>
>         <param name="username" value="071234"/>
>         <param name="password" value="1234"/>
>         <param name="register-proxy" value="118.69.145.5"/>
>         <param name="register" value="true"/>
>         <param name="retry-seconds" value="10"/>
>     </gateway>
> </include>
>
> so i use xlite register to freeswitch with user 1000
> i start freeswitch , freeswitch start ok,
> freeswitch register to sip server successfull with username 071234
> xlite register to freeswitch successfull
>
> so i edit the public.xml
> for incomming call from sip server, i will forward to extension 1000
>     <extension name="sip.yeah.com">
>       <condition field="destination_number" expression="071234">
>         <action application="transfer" data="1000 XML default"/>
>       </condition>
>     </extension>
>
> but when i use xlite make call out from freeswitch to sip server and  
> get no routing
>
> nta: received INVITE sip:071234 at 192.168.1.90;user=phone SIP/2.0  
> (CSeq 3621)
> nta: INVITE (3621) going to a default leg
> nua(0xb6e20f90): adding session usage
> nta: sent 100 Trying for INVITE (3621)
> nua(0xb6e20f90): event i_invite 100 Trying
> nua(0xb6e20f90): call state changed: init -> received, received offer
> nua(0xb6e20f90): event i_state 100 Trying
> 432146 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.90 
>  [ef1af4b1-4d07-4b8e-8520-d99363347006]
> 446099 [INFO] mod_dialplan_xml.c:418 Processing 1000->071234 in  
> context default
> 455113 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 192.168.1.90 
>  to enum[071234 at default]
> 566021 [INFO] switch_core_state_machine.c:142 No Route, Aborting
> 566021 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 192.168.1.90 
>  [CS_ROUTING] [NO_ROUTE_DESTINATION]
> nua(0xb6e20f90): recv signal r_respond 404 Not Found
>
> i am not clear about routing call in freeswitch, i did read the  
> dialplan in wiki but still not clear, is there any info to easy  
> understand routing in freeswitch
>
> please help,
>
> Thank you
>
>
> L??t web nhanh h?n
> Internet Explorer 8 t?i ?u h?a cho Yahoo!, t? ??ng kh?i  
> ??ng 2 trang b?n th?ch m?i l?n m? tr?nh duy?t. T?i  
> IE8 t?i ??y! (Mi?n  
> ph?)_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From cucku.cucku at yahoo.com.vn  Tue Mar 30 00:01:55 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Tue, 30 Mar 2010 15:01:55 +0800 (SGT)
Subject: [Freeswitch-users] need help on call to gateway
In-Reply-To: <5A16ECE3-D398-4EA8-82CD-C09E6C3EB897@freeswitch.org>
Message-ID: <319878.5002.qm@web76204.mail.sg1.yahoo.com>

Hi Brian

could you guide me more to config on freeswitch

i want xilte with user 1000 can make call out to sip server 
and the prefix when make call out to sip server : 0
and receive incomming call from SIP server

what file name i should create 

Thank you


--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:

T?: Brian West <brian at freeswitch.org>
Ch? ??: Re: [Freeswitch-users] need help on call to gateway
??n: freeswitch-users at lists.freeswitch.org
Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37

its looking in context default... not public.
/b
On Mar 29, 2010, at 11:33 PM, false wrote:
Hi all

i use freeswitch to register to sip server
the domain is not real: sip.yeah.com
the outbound proxy : 118.69.145.5

i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
</include>
??? <gateway name="sip.yeah.com">
??? ??? <param name="username" value="071234"/>
??? ??? <param name="from-user" value="071234"/>??? 
??? ??? <param name="proxy" value="sip.yeah.com"/>
??? ??? <param name="username" value="071234"/>
??? ??? <param name="password" value="1234"/>
??? ??? <param name="register-proxy"
 value="118.69.145.5"/>
??? ??? <param name="register" value="true"/>
??? ??? <param name="retry-seconds" value="10"/>
??? </gateway>
</include>

so i use xlite register to freeswitch with user 1000
i start freeswitch , freeswitch start ok,
freeswitch register to sip server successfull with username 071234
xlite register to freeswitch successfull

so i edit the public.xml
for incomming call from sip server, i will forward to extension 1000
??? <extension name="sip.yeah.com">
????? <condition field="destination_number" expression="071234">
??????? <action application="transfer" data="1000 XML default"/>
????? </condition>
??? </extension>

but when i use
 xlite make call out from freeswitch to sip server and get no routing




      ____________________________________________________________
T?i sao ??n ?ng l?i th?ch nh?n ph? n? c? m?i t?c d?i h?n?
Kh?m ph? t?i Yahoo! H?i & ??p
http://vn.answers.yahoo.com
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From maciej.aniserowicz at gmail.com  Tue Mar 30 00:06:50 2010
From: maciej.aniserowicz at gmail.com (Maciej Aniserowicz)
Date: Tue, 30 Mar 2010 09:06:50 +0200
Subject: [Freeswitch-users] Error when recording
In-Reply-To: <736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
References: <1269856895179-4817081.post@n2.nabble.com>
	<736D3688-7DAF-4EF0-AB66-E6D3A4DCB02F@freeswitch.org>
Message-ID: <4BB1A30A.4060701@gmail.com>

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From david.ponzone at gmail.com  Tue Mar 30 00:26:13 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 09:26:13 +0200
Subject: [Freeswitch-users] need help on call to gateway
In-Reply-To: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
Message-ID: <FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>

False,

I think Brian is sleeping so I give you a quick answer.

For OUTBOUND:

Following my previous mail, you have to check if your user 1000 has  
the default context configured.
If it does, then you just need to add a file named 01_whatever.xml in  
conf/dialplan/default/.
(You may also remove from this path the remaining .xml files from the  
default install)
In 01_whatever.xml, put:

<include>
     <extension name="Route_to_sip.yeah.com">
       <condition field="destination_number" expression="^(\d+)$">
         <action application="bridge" data="sofia/gateway/sip.yeah.com/ 
0$1"/>
       </condition>
     </extension>
<include>

So when you dial any number not previously matched in default.xml  
(local users, voicemail, ...), this will bridge it to your gateway,  
adding 0 as a prefix.

For INBOUND:

In conf/dialplan/public/, add a file named whatever.xml with the  
following content:

<include>
    <extension name="Inbound_from_sip.yeah.com">
       <condition field="destination_number" expression="^YOUR_DID$">
         <action application="transfer" data="1000 XML default"/>
       </condition>
     </extension>
</include>

Here, YOUR_DID is the DID number your provider allocated to you.
That is an exact match, so if you're not sure about the exact format,  
remove the leading ^, and match that on the longest part of the number  
you know about.

Of course, after doing all this, reloadxml.

It will perhaps not fit in your config exactly, but you should get the  
idea to get started.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 30/03/2010 ? 09:01, false a ?crit :

> Hi Brian
>
> could you guide me more to config on freeswitch
>
> i want xilte with user 1000 can make call out to sip server
> and the prefix when make call out to sip server : 0
> and receive incomming call from SIP server
>
> what file name i should create
>
> Thank you
>
>
> --- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ??  
> vi?t:
>
> T?: Brian West <brian at freeswitch.org>
> Ch? ??: Re: [Freeswitch-users] need help on call to gateway
> ??n: freeswitch-users at lists.freeswitch.org
> Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>
> its looking in context default... not public.
>
> /b
>
> On Mar 29, 2010, at 11:33 PM, false wrote:
>
>> Hi all
>>
>> i use freeswitch to register to sip server
>> the domain is not real: sip.yeah.com
>> the outbound proxy : 118.69.145.5
>>
>> i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/ 
>> sip_profiles/external folder
>> </include>
>>     <gateway name="sip.yeah.com">
>>         <param name="username" value="071234"/>
>>         <param name="from-user" value="071234"/>
>>         <param name="proxy" value="sip.yeah.com"/>
>>         <param name="username" value="071234"/>
>>         <param name="password" value="1234"/>
>>         <param name="register-proxy" value="118.69.145.5"/>
>>         <param name="register" value="true"/>
>>         <param name="retry-seconds" value="10"/>
>>     </gateway>
>> </include>
>>
>> so i use xlite register to freeswitch with user 1000
>> i start freeswitch , freeswitch start ok,
>> freeswitch register to sip server successfull with username 071234
>> xlite register to freeswitch successfull
>>
>> so i edit the public.xml
>> for incomming call from sip server, i will forward to extension 1000
>>     <extension name="sip.yeah.com">
>>       <condition field="destination_number" expression="071234">
>>         <action application="transfer" data="1000 XML default"/>
>>       </condition>
>>     </extension>
>>
>> but when i use xlite make call out from freeswitch to sip server  
>> and get no routing
>
>
> T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo!  
> Mail m?i h?m nay! _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From siniypin at gmail.com  Tue Mar 30 02:35:32 2010
From: siniypin at gmail.com (RobertT)
Date: Tue, 30 Mar 2010 13:35:32 +0400
Subject: [Freeswitch-users] Experience with FreeSWITCH & Turnserver.org ?
Message-ID: <2160023e1003300235i24b5d32apb7e0be71295ff018@mail.gmail.com>

I chose this kind of setup, because I wanted to enforce ICE session
negotiation between clients, that could possibly call from various kinds of
NATs.
sip_nat_detected tells you if the caller is residing behind NAT, but you
don't know anything about another party yet. It might be hidden with NAT as
well.

Best regards, RobertT.
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From hungngm at bkav.com.vn  Tue Mar 30 03:46:50 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Tue, 30 Mar 2010 17:46:50 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <5369E7454703C42D03169AF474CC4B772E713FFF@hungngm>



 

Hi SeVen Du.

Thank for your reply.

About question 2: I think in real life (call center), supporters in queue must
be known the real Caller_ID_number of customers. If caller_id_number is replated
by 'fifo+<fifo_name>', caller_id_number will be invaluable informations.

So, Is there anyway to preserve Caller_ID_number from the invaluable change as
origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'.

Best regard !

> Hi all.
>
> I am newbegin and have 2 question as follow
>
> 1. Can config mod_fifo with member in external register ???
>
> In my lab, i have OpenSIP as Proxy server and Register server and
> Freeswitch as media server config mod_fifo (queue)

>
> User with extension 1010 is registerd with OpenSIPS server, queue 8000 is
> media service in freeswitch server. Can i config mod_fifo in freeswitch
> with member in Opensips server ?
>
> I try config in /fifo.conf.xml like
>
> > lag="5">{fifo_member_wait=nowait}user/1010 at opensips_IP
>
> but it not work.
>

because it's not a valid dialstring. try

sofia/ / instead of user/

it's only a dialstring, so, if you can make "originate
" work then fifo should work.

> Someone done this with opensips or another register server, please post some
> guide.
>
>
>
> 2. With mod_fifo: caller_id_name and caller_id_number will be replated by
> default config:
>
>
origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'.
> So, if caller_id_number is 0123456.. it will be replates wit
> 'fifo+test_fifo'. I want origination_caller_id_number has value is
> 0123456..
>
no, because you don't know which caller will actually been bridged to
which agent before the agent actually answer the call.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mike at jerris.com  Tue Mar 30 04:10:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 30 Mar 2010 07:10:10 -0400
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <5369E7454703C42D03169AF474CC4B772E713FFF@hungngm>
References: <5369E7454703C42D03169AF474CC4B772E713FFF@hungngm>
Message-ID: <0537B750-D1EF-47B6-9C7D-FB510429CD48@jerris.com>


On Mar 30, 2010, at 6:46 AM, Nguy?n M?nh H?ng wrote:

> Hi SeVen Du.
> Thank for your reply.
>  
> About question 2: I think in real life (call center), supporters in queue must be known the real Caller_ID_number of  customers. If caller_id_number is replated by 'fifo+<fifo_name>', caller_id_number will be invaluable informations.
> So, Is there anyway to preserve Caller_ID_number from the invaluable change as origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'.

More information in the following thread:

http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050153.html

You can't know the caller id number when the call has been placed because it is not yet decided until the call is answered. 

Mike


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From abid_freeswitch at live.com  Tue Mar 30 04:52:07 2010
From: abid_freeswitch at live.com (Abid Saleem)
Date: Tue, 30 Mar 2010 16:52:07 +0500
Subject: [Freeswitch-users] Radius/Authentication/Authorization
Message-ID: <COL109-W15576D4B6E4D7BFFFFEDD7F41F0@phx.gbl>


Hi Neal and other Contributors to FS,
I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration.
If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation.
Regards----------Abid SaleemSr. Product Manager
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From velu.technical at gmail.com  Tue Mar 30 05:16:32 2010
From: velu.technical at gmail.com (velusamy Krishnan)
Date: Tue, 30 Mar 2010 17:46:32 +0530
Subject: [Freeswitch-users] Invalid Boost Version 103 Expecting 100
In-Reply-To: <c4d05cbe1003250658ue743dd4yfd4be2062884c1d7@mail.gmail.com>
References: <1452e2981003240429x510bec4k3a57beb58d703eb8@mail.gmail.com>
	<191c3a031003240758k1e28b8c2vd356485e4a96d45c@mail.gmail.com>
	<1452e2981003242152l2994d0eai49a1004654095ee4@mail.gmail.com>
	<c4d05cbe1003250658ue743dd4yfd4be2062884c1d7@mail.gmail.com>
Message-ID: <1452e2981003300516s4d684778xb920cb64424d469d@mail.gmail.com>

Updated the freeswitch latest SVN trunk (17127). Installed the latest
wanpipe drivers.
        Sangoma PRI Protocol Stack Daemon
        Version: 1.75
        Date: Mar 18 2010
        Wanpipe Release: wanpipe-3.5.10.10
        Revision:Revision: 15921

Now it is working fine. Thanks for your valuable guidelines..


On Thu, Mar 25, 2010 at 7:28 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Actually the log message indicates you have older openzap and newer wanpipe
> (OpenZAP expects 100 and is receiving 103, therefore OpenZAP is the outdated
> here).
>
> Update to latest openzap.
>
>
> On Thu, Mar 25, 2010 at 12:52 AM, velusamy Krishnan <
> velu.technical at gmail.com> wrote:
>
>> Installed the wanpipe version(wanpipe-3.5.10.10b.tgz) which is suggested
>> in the http://wiki.sangoma.com/wanpipe-freeswitch-install.
>>
>> But, the same is happening.
>> >From sangoma_mgd.log file found the following,
>>
>> sangoma_prid: = Sangoma PRI Protocol Stack Daemon
>> sangoma_prid: = Version: 1.75
>> sangoma_prid: = Date: Mar 18 2010
>> sangoma_prid: = Wanpipe Release: wanpipe-3.5.10.10
>> sangoma_prid: = Revision:Revision: 15921
>>
>> Do you mean some other wanpipe release?
>>
>> On Wed, Mar 24, 2010 at 8:28 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> you need the newest beta revision of wanpipe to use with latest trunk
>>>
>>> On Wed, Mar 24, 2010 at 6:29 AM, velusamy Krishnan <
>>> velu.technical at gmail.com> wrote:
>>>
>>>> Dear all,
>>>>
>>>>     Compiled the wanpipe with freeswitch and configured it.
>>>>     When I originate a call from the freeswitch command line it said the
>>>> following error and got core dump.
>>>>
>>>>  freeswitch at IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753
>>>> &park()
>>>> 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT
>>>> (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0
>>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid
>>>> Boost Version 103  Expecting 100
>>>> 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical
>>>> Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027
>>>>
>>>> NOTE:  freeswitch  version is  1.0.trunk ( 17084)
>>>>               wanpipe version is 3.5.8.6.
>>>>               smg_prid version is 1.63
>>>>
>>>> Please help me.........
>>>>
>>>> Thanks..
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Tue Mar 30 09:14:29 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 30 Mar 2010 09:14:29 -0700
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com>
Message-ID: <e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com>

Here is the log output (with names/IPs hidden) from my broken server
executing my lua app:

EXECUTE sofia/internal/1001 at broken.server.eduhash(insert/broken.server.IP-spymap/1001/e93b00d7-fb0f-413b-9f17-207ea444239a)
EXECUTE sofia/internal/1001 at broken.server.eduhash(insert/broken.server.IP-last_dial/1001/5000)
EXECUTE sofia/internal/1001 at broken.server.eduhash(insert/broken.server.IP-last_dial/global/e93b00d7-fb0f-413b-9f17-207ea444239a)
EXECUTE sofia/internal/1001 at broken.server.edu lua(AO/otalo.lua)
2010-03-30 08:49:24.092796 [INFO] switch_cpp.cpp:1142 otalo.lua : caller id
= 1001
2010-03-30 08:49:24.093798 [INFO] switch_cpp.cpp:1142 otalo.lua : user id =
1
2010-03-30 08:49:24.099786 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/internal/1001 at broken.server.edu] broken.server.IP port 26884 ->
127.0.0.1 port 4000 codec: 3 ms: 20
2010-03-30 08:49:24.099786 [DEBUG] switch_rtp.c:1182 Starting timer [soft]
160 bytes per 20ms
2010-03-30 08:49:24.101833 [DEBUG] mod_sofia.c:636 Local SDP sofia/internal/
1001 at broken.server.edu:
v=0
o=FreeSWITCH 1269937280 1269937281 IN IP4 broken.server.IP
s=FreeSWITCH
c=IN IP4 broken.server.IP
t=0 0
m=audio 26884 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-03-30 08:49:24.101833 [DEBUG] switch_core_session.c:638 Send signal
sofia/internal/1001 at broken.server.edu [BREAK]
2010-03-30 08:49:24.101833 [NOTICE] switch_cpp.cpp:554 Channel
[sofia/internal/1001 at broken.server.edu] has been answered
2010-03-30 08:49:24.101833 [DEBUG] sofia.c:4152 Channel sofia/internal/
1001 at broken.server.edu entering state [completed][200]
2010-03-30 08:49:24.146888 [DEBUG] sofia.c:4152 Channel sofia/internal/
1001 at broken.server.edu entering state [ready][200]
2010-03-30 08:49:25.115874 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:44.335874 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:44.835892 [INFO] switch_cpp.cpp:1142 otalo.lua : SELECT id,
name_file FROM app_tbl ORDER BY id ASC
2010-03-30 08:49:44.835892 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:44.835892 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:44.855887 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:46.355889 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:46.375877 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:49.495889 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:49.515889 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:50.215892 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:50.735890 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:50.735890 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:50.755890 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 08:49:52.515889 [DEBUG] switch_ivr_play_say.c:1444 done playing
file
2010-03-30 08:49:52.535889 [DEBUG] switch_ivr_play_say.c:1152 Codec
Activated L16 at 8000hz 1 channels 20ms

And here is pretty much up to the same point in the app executed on my
older, working server:

EXECUTE sofia/internal/1001 at working.server.eduhash(insert/working.server.edu-spymap/1001/e8b73a98-b783-4c73-8547-2e2c20786b0d)
EXECUTE sofia/internal/1001 at working.server.eduhash(insert/working.server.edu-last_dial/1001/5000)
EXECUTE sofia/internal/1001 at working.server.eduhash(insert/working.server.edu-last_dial/global/e8b73a98-b783-4c73-8547-2e2c20786b0d)
EXECUTE sofia/internal/1001 at working.server.edu lua(AO/otalo.lua)
2010-03-30 21:20:51.763507 [DEBUG] switch_cpp.cpp:554 sofia/internal/
1001 at working.server.edu receive message [ANSWER]
2010-03-30 21:20:51.763507 [DEBUG] sofia_glue.c:2381 AUDIO RTP
[sofia/internal/1001 at working.server.edu] working.server.edu port 32338 ->
127.0.0.1 port 4002 codec: 3 ms: 20
2010-03-30 21:20:51.763507 [DEBUG] switch_rtp.c:1167 Starting timer [soft]
160 bytes per 20ms
2010-03-30 21:20:51.764569 [DEBUG] mod_sofia.c:571 Local SDP sofia/internal/
1001 at working.server.edu:
v=0
o=FreeSWITCH 1269931913 1269931914 IN IP4 working.server.edu
s=FreeSWITCH
c=IN IP4 working.server.edu
t=0 0
m=audio 32338 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2010-03-30 21:20:51.765633 [DEBUG] switch_core_session.c:645 Send signal
sofia/internal/1001 at working.server.edu [BREAK]
2010-03-30 21:20:51.765633 [DEBUG] sofia.c:3774 Channel sofia/internal/
1001 at working.server.edu entering state [completed][200]
2010-03-30 21:20:51.765633 [NOTICE] switch_cpp.cpp:554 Channel
[sofia/internal/1001 at working.server.edu] has been answered
2010-03-30 21:20:51.765633 [DEBUG] switch_channel.c:182 sofia/internal/
1001 at working.server.edu receive message [AUDIO_SYNC]
2010-03-30 21:20:52.132157 [DEBUG] sofia.c:3774 Channel sofia/internal/
1001 at working.server.edu entering state [ready][200]
2010-03-30 21:20:52.346042 [INFO] switch_rtp.c:1987 Auto Changing port from
127.0.0.1:4002 to 75.42.72.134:4002
2010-03-30 21:20:52.766898 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:20:52.766898 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:11.985960 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:12.486077 [INFO] switch_cpp.cpp:1129 otalo.lua : SELECT id,
name_file FROM app_tbl ORDER BY id ASC
2010-03-30 21:21:12.486077 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:12.486077 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:12.506154 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:12.506154 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:14.006120 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:14.026376 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:14.026376 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:17.146408 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:17.166614 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:17.166614 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:17.866526 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:18.366651 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:18.366651 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:18.386040 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:18.386040 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:20.146593 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:20.166701 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:20.166701 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:23.286448 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:23.306606 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:23.306606 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:24.126360 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:24.646291 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:24.646291 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:24.666736 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:24.666736 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:26.166139 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:26.186290 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:26.186290 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:29.306221 [DEBUG] switch_ivr_play_say.c:1446 done playing
file
2010-03-30 21:21:29.326378 [DEBUG] switch_ivr_play_say.c:1154 Codec
Activated L16 at 8000hz 1 channels 20ms
2010-03-30 21:21:29.326378 [DEBUG] switch_core_io.c:652 sofia/internal/
1001 at working.server.edu receive message [TRANSCODING_NECESSARY]
2010-03-30 21:21:30.126585 [DEBUG] switch_ivr_play_say.c:1446 done playing
file


I think the most significantly the broken server doesn't give the  '...
receive message [TRANSCODING_NECESSARY]' output. The new server is Ubuntu
9.0.4 running FS downloaded from main trunk in Dec 09, the broken is CentOS
5.4 running FS main trunk downloaded this month (not sure how to get version
info from FS).

Let me know if more info is necessary to solve this, and thanks again!
Neil

On Mon, Mar 29, 2010 at 9:19 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> oops, sorry about that. Here's read:
>
> function read(file, delay)
>    if (digits == "") then
>       logfile:write(sessid, "\t",
>     session:getVariable("caller_id_number"), "\t",
>     os.time(), "\t", "Prompt", "\t", file, "\n");
>       digits = session:read(1, 1, file, delay, "#");
>       if (digits ~= "") then
>  logfile:write(sessid, "\t", session:getVariable("caller_id_number"), "\t",
> os.time(), "\t", "dtmf", "\t", file, "\t", digits, "\n");
>       end
>    end
> end
>
> On Mon, Mar 29, 2010 at 8:28 PM, Brian West <brian at freeswitch.org> wrote:
>
>> what is this code?
>>
>> /b
>>
>> On Mar 29, 2010, at 10:21 PM, Neil Patel wrote:
>>
>> > read(absolute_path_to_wav, 0);
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From william.suffill at gmail.com  Tue Mar 30 09:26:55 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 30 Mar 2010 12:26:55 -0400
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com> 
	<e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com>
Message-ID: <6b65470d1003300926u3e85555ej942c84d021a7dbb7@mail.gmail.com>

in fs_cli run version.

For Example:
fs_cli -x "version"
FreeSWITCH Version 1.0.4 (exported)
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From william.suffill at gmail.com  Tue Mar 30 09:27:42 2010
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 30 Mar 2010 12:27:42 -0400
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <6b65470d1003300926u3e85555ej942c84d021a7dbb7@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com> 
	<e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com> 
	<6b65470d1003300926u3e85555ej942c84d021a7dbb7@mail.gmail.com>
Message-ID: <6b65470d1003300927v7646585bh7c4e231dd7a86dcd@mail.gmail.com>

Running fs_cli then running version also works.
--
freeswitch at internal> version
FreeSWITCH Version 1.0.4 (exported)
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From lists at infosecurity.ch  Tue Mar 30 09:36:51 2010
From: lists at infosecurity.ch (Fabio Pietrosanti (naif))
Date: Tue, 30 Mar 2010 18:36:51 +0200
Subject: [Freeswitch-users] ringback with unsupported codec
Message-ID: <4BB228A3.8030007@infosecurity.ch>

Hi all,

i would like to provide ringback in early media using AMR that's 
supported only in passtrough.

Would it possible, providing static AMR sample to be played back, to 
play it?

Fabio Pietrosanti


From brian at freeswitch.org  Tue Mar 30 09:41:53 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Mar 2010 11:41:53 -0500
Subject: [Freeswitch-users] ringback with unsupported codec
In-Reply-To: <4BB228A3.8030007@infosecurity.ch>
References: <4BB228A3.8030007@infosecurity.ch>
Message-ID: <600DE716-2F00-4CB0-B32A-C600B0912B56@freeswitch.org>

Have you tried?  I recall it should work.

/b

On Mar 30, 2010, at 11:36 AM, Fabio Pietrosanti (naif) wrote:

> Would it possible, providing static AMR sample to be played back, to 
> play it?



From dome at tel.co.th  Tue Mar 30 09:49:43 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 30 Mar 2010 23:49:43 +0700
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
Message-ID: <8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>

Thanks All.
How to deny all and allow some ?
I try
    <list name="test" default="allow">
        <node type="deny" cidr="0.0.0.0/0"/>
        <node type="allow" cidr="10.0.0.1/32"/>
        <node type="allow" cidr="115.178.1.0/24"/>
    </list>

but don't work.

Dome C.

2010/3/30 Madovsky <infos at madovsky.org>:
> you need to use ACL in autoload_configs/acl.conf.xml
>
> ----- Original Message -----
> From: "Dome Charoenyost" <dome at tel.co.th>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, March 29, 2010 3:05 PM
> Subject: [Freeswitch-users] Regular Expression for match ip address
>
>
>> Dear All,
>> ? I want to autorize incoming call to public context by IP by
>> ? ? <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>> but some time i want to allow 1 Class C or some range of IP. How can do
>> that ?
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Tue Mar 30 09:59:14 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 30 Mar 2010 23:59:14 +0700
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
Message-ID: <8ccbff061003300959j4dbc1942te43b738bffe90a80@mail.gmail.com>

Got it :)
    <list name="test" default="deny">
        <node type="allow" cidr="10.0.0.1/32"/>
        <node type="allow" cidr="115.178.1.0/24"/>
    </list>



2010/3/30 Dome Charoenyost <dome at tel.co.th>:
> Thanks All.
> How to deny all and allow some ?
> I try
> ? ?<list name="test" default="allow">
> ? ? ? ?<node type="deny" cidr="0.0.0.0/0"/>
> ? ? ? ?<node type="allow" cidr="10.0.0.1/32"/>
> ? ? ? ?<node type="allow" cidr="115.178.1.0/24"/>
> ? ?</list>
>
> but don't work.
>
> Dome C.
>
> 2010/3/30 Madovsky <infos at madovsky.org>:
>> you need to use ACL in autoload_configs/acl.conf.xml
>>
>> ----- Original Message -----
>> From: "Dome Charoenyost" <dome at tel.co.th>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 29, 2010 3:05 PM
>> Subject: [Freeswitch-users] Regular Expression for match ip address
>>
>>
>>> Dear All,
>>> ? I want to autorize incoming call to public context by IP by
>>> ? ? <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>>> but some time i want to allow 1 Class C or some range of IP. How can do
>>> that ?
>>>
>>> BG
>>>
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From rob4manhere at gmail.com  Tue Mar 30 10:01:54 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 30 Mar 2010 12:01:54 -0500
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
Message-ID: <e15d328c1003301001m40460da4n444ca710619d92bb@mail.gmail.com>

Have you tried changing the default?

<list name="test" *default="allow"*>


On Tue, Mar 30, 2010 at 11:49 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Thanks All.
> How to deny all and allow some ?
> I try
>    <list name="test" default="allow">
>        <node type="deny" cidr="0.0.0.0/0"/>
>        <node type="allow" cidr="10.0.0.1/32"/>
>        <node type="allow" cidr="115.178.1.0/24"/>
>    </list>
>
> but don't work.
>
> Dome C.
>
> 2010/3/30 Madovsky <infos at madovsky.org>:
> > you need to use ACL in autoload_configs/acl.conf.xml
> >
> > ----- Original Message -----
> > From: "Dome Charoenyost" <dome at tel.co.th>
> > To: <freeswitch-users at lists.freeswitch.org>
> > Sent: Monday, March 29, 2010 3:05 PM
> > Subject: [Freeswitch-users] Regular Expression for match ip address
> >
> >
> >> Dear All,
> >>   I want to autorize incoming call to public context by IP by
> >>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> >> but some time i want to allow 1 Class C or some range of IP. How can do
> >> that ?
> >>
> >> BG
> >>
> >> Dome C.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rob4manhere at gmail.com  Tue Mar 30 10:05:47 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 30 Mar 2010 12:05:47 -0500
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003300959j4dbc1942te43b738bffe90a80@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
	<8ccbff061003300959j4dbc1942te43b738bffe90a80@mail.gmail.com>
Message-ID: <e15d328c1003301005k5f793345jcabcd06296a15e65@mail.gmail.com>

Figured you would ;-)

On Tue, Mar 30, 2010 at 11:59 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Got it :)
>    <list name="test" default="deny">
>         <node type="allow" cidr="10.0.0.1/32"/>
>        <node type="allow" cidr="115.178.1.0/24"/>
>    </list>
>
>
>
> 2010/3/30 Dome Charoenyost <dome at tel.co.th>:
> > Thanks All.
> > How to deny all and allow some ?
> > I try
> >    <list name="test" default="allow">
> >        <node type="deny" cidr="0.0.0.0/0"/>
> >        <node type="allow" cidr="10.0.0.1/32"/>
> >        <node type="allow" cidr="115.178.1.0/24"/>
> >    </list>
> >
> > but don't work.
> >
> > Dome C.
> >
> > 2010/3/30 Madovsky <infos at madovsky.org>:
> >> you need to use ACL in autoload_configs/acl.conf.xml
> >>
> >> ----- Original Message -----
> >> From: "Dome Charoenyost" <dome at tel.co.th>
> >> To: <freeswitch-users at lists.freeswitch.org>
> >> Sent: Monday, March 29, 2010 3:05 PM
> >> Subject: [Freeswitch-users] Regular Expression for match ip address
> >>
> >>
> >>> Dear All,
> >>>   I want to autorize incoming call to public context by IP by
> >>>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> >>> but some time i want to allow 1 Class C or some range of IP. How can do
> >>> that ?
> >>>
> >>> BG
> >>>
> >>> Dome C.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rob4manhere at gmail.com  Tue Mar 30 10:05:47 2010
From: rob4manhere at gmail.com (Rob Forman)
Date: Tue, 30 Mar 2010 12:05:47 -0500
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <8ccbff061003300959j4dbc1942te43b738bffe90a80@mail.gmail.com>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
	<8ccbff061003300959j4dbc1942te43b738bffe90a80@mail.gmail.com>
Message-ID: <e15d328c1003301005k5f793345jcabcd06296a15e65@mail.gmail.com>

Figured you would ;-)

On Tue, Mar 30, 2010 at 11:59 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Got it :)
>    <list name="test" default="deny">
>         <node type="allow" cidr="10.0.0.1/32"/>
>        <node type="allow" cidr="115.178.1.0/24"/>
>    </list>
>
>
>
> 2010/3/30 Dome Charoenyost <dome at tel.co.th>:
> > Thanks All.
> > How to deny all and allow some ?
> > I try
> >    <list name="test" default="allow">
> >        <node type="deny" cidr="0.0.0.0/0"/>
> >        <node type="allow" cidr="10.0.0.1/32"/>
> >        <node type="allow" cidr="115.178.1.0/24"/>
> >    </list>
> >
> > but don't work.
> >
> > Dome C.
> >
> > 2010/3/30 Madovsky <infos at madovsky.org>:
> >> you need to use ACL in autoload_configs/acl.conf.xml
> >>
> >> ----- Original Message -----
> >> From: "Dome Charoenyost" <dome at tel.co.th>
> >> To: <freeswitch-users at lists.freeswitch.org>
> >> Sent: Monday, March 29, 2010 3:05 PM
> >> Subject: [Freeswitch-users] Regular Expression for match ip address
> >>
> >>
> >>> Dear All,
> >>>   I want to autorize incoming call to public context by IP by
> >>>     <condition field="network_addr" expression="^192\.168\.1\.1$"/>
> >>> but some time i want to allow 1 Class C or some range of IP. How can do
> >>> that ?
> >>>
> >>> BG
> >>>
> >>> Dome C.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sean at obscuradigital.com  Tue Mar 30 10:21:14 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 10:21:14 -0700
Subject: [Freeswitch-users] Bandwidth.com
In-Reply-To: <ec2ce17e1003291901q7a280748v7fcc57ded2d3466c@mail.gmail.com>
Message-ID: <C7D7811A.7770%sean@obscuradigital.com>

That worked thanks.

Sean


On 3/29/10 7:01 PM, "jay binks" <jaybinks at gmail.com> wrote:

> I believe you need to still specify the username and password... but
> use a "dummy" string.
> from memory, they just get ignored but doing this satisfies freeswitch.
> 
> 
> On Tue, Mar 30, 2010 at 11:49 AM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello,
>> 
>> Just received my bandwidth.com provisioning email with my sip_trunks and DID
>> numbers. ?However, part of the instructions say I can?t use a username and
>> password when connecting to the gateway. ?I tried leaving the username and
>> password blank in the gateway sip profile but it keeps giving me an error
>> message, Error: username param is REQUIRED. ?Anyone ever setup a gateway
>> with Bandwidth and how does one get around the username param requirement.
>> 
>> Thanks
>> Sean
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 




From anthony.minessale at gmail.com  Tue Mar 30 10:26:14 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 30 Mar 2010 12:26:14 -0500
Subject: [Freeswitch-users] ringback with unsupported codec
In-Reply-To: <600DE716-2F00-4CB0-B32A-C600B0912B56@freeswitch.org>
References: <4BB228A3.8030007@infosecurity.ch>
	<600DE716-2F00-4CB0-B32A-C600B0912B56@freeswitch.org>
Message-ID: <191c3a031003301026j1b405879h450ac2c568335c4c@mail.gmail.com>

record the file and use mod_native_file to play it by omitting the file
extension

set ringback=foo  to imply foo.AMR


On Tue, Mar 30, 2010 at 11:41 AM, Brian West <brian at freeswitch.org> wrote:

> Have you tried?  I recall it should work.
>
> /b
>
> On Mar 30, 2010, at 11:36 AM, Fabio Pietrosanti (naif) wrote:
>
> > Would it possible, providing static AMR sample to be played back, to
> > play it?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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From troy at tlainvestments.com  Tue Mar 30 10:41:31 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 30 Mar 2010 10:41:31 -0700
Subject: [Freeswitch-users] Cisco 7940 To Private
Message-ID: <51FE9B78-4CBE-4352-B250-B76F4CDD8B3B@tlainvestments.com>

I have some Cisco 7940's registered and working well with FS, but whenever I dial out from the Cisco phone, it indicates the call is "To Private" rather than showing the number I'm calling or something.  I've put the SIP trace in pastebin http://pastebin.freeswitch.org/12589 - there is a Remote-Part-ID field that seems OK.  Does anyone know what Cisco is looking for here?

Thanks,
Troy




From brian at freeswitch.org  Tue Mar 30 10:53:21 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Mar 2010 12:53:21 -0500
Subject: [Freeswitch-users] Cisco 7940 To Private
In-Reply-To: <51FE9B78-4CBE-4352-B250-B76F4CDD8B3B@tlainvestments.com>
References: <51FE9B78-4CBE-4352-B250-B76F4CDD8B3B@tlainvestments.com>
Message-ID: <E151C0A3-250B-43BB-A399-C54D8B71137D@freeswitch.org>

svn update this is fixed already.

/b

On Mar 30, 2010, at 12:41 PM, Troy Anderson wrote:

> I have some Cisco 7940's registered and working well with FS, but whenever I dial out from the Cisco phone, it indicates the call is "To Private" rather than showing the number I'm calling or something.  I've put the SIP trace in pastebin http://pastebin.freeswitch.org/12589 - there is a Remote-Part-ID field that seems OK.  Does anyone know what Cisco is looking for here?
> 
> Thanks,
> Troy

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From sean at obscuradigital.com  Tue Mar 30 11:01:12 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 11:01:12 -0700
Subject: [Freeswitch-users] DTMF
Message-ID: <C7D78A78.7775%sean@obscuradigital.com>

Hello,

I?ve posted on the question on this topic before but I?ll try again with
more detail.

I have this svn version
FreeSWITCH Version 1.0.trunk (16952)
SIP provider: Bandwidth.com

I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
recognize the incoming call tones.  Bandwidth can set up either inband or
RC2833.  They require these codecs, G.711Ulaw and G.729A.

The tech at Bandwidth wanted me to setup inband to test if that works
better, but wasn?t sure where to set it up.  I added this to my dialplan

<extension name="main_number_23" continue="true">
      <condition field="destination_number" expression="^3000$">
        <action application="answer"/>
        <action application="start_dtmf"/>
        <action application="sleep" data="1000"/>
        <action application="conference" data="conference_1 at conference_1"/>
      </condition>
</extension>

Should I add it to my sip trunk?

Not sure how to troubleshoot this issue

Thanks
Sean
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From infos at madovsky.org  Tue Mar 30 10:06:42 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 30 Mar 2010 13:06:42 -0400
Subject: [Freeswitch-users] Regular Expression for match ip address
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com><9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
Message-ID: <B0DB56DE1A764F10AB32AB2CDD1A2D99@MOBILEE1705>

if you want this
<node type="deny" cidr="0.0.0.0/0"/>

so use <list name="test" default="deny">
and use allow inside

----- Original Message ----- 
From: "Dome Charoenyost" <dome at tel.co.th>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, March 30, 2010 12:49 PM
Subject: Re: [Freeswitch-users] Regular Expression for match ip address


> Thanks All.
> How to deny all and allow some ?
> I try
>    <list name="test" default="allow">
>        <node type="deny" cidr="0.0.0.0/0"/>
>        <node type="allow" cidr="10.0.0.1/32"/>
>        <node type="allow" cidr="115.178.1.0/24"/>
>    </list>
>
> but don't work.
>
> Dome C.
>
> 2010/3/30 Madovsky <infos at madovsky.org>:
>> you need to use ACL in autoload_configs/acl.conf.xml
>>
>> ----- Original Message -----
>> From: "Dome Charoenyost" <dome at tel.co.th>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, March 29, 2010 3:05 PM
>> Subject: [Freeswitch-users] Regular Expression for match ip address
>>
>>
>>> Dear All,
>>> I want to autorize incoming call to public context by IP by
>>> <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>>> but some time i want to allow 1 Class C or some range of IP. How can do
>>> that ?
>>>
>>> BG
>>>
>>> Dome C.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From infos at madovsky.org  Tue Mar 30 10:18:13 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 30 Mar 2010 13:18:13 -0400
Subject: [Freeswitch-users] domain name in shared DB
Message-ID: <9099D7E46BBB410B893CF5BEC23C3C26@MOBILEE1705>

2 nodes in shared DB, node1 and node2 respectively with hostname node1.mydomain.org node2.mydomain.org
the profile name is called mydomain.org which is shared by the 2 nodes.
a user registers on node1 and wait calls.
someone from external gateway call him, so user at node1.mydomain.org or user at node2.mydomain.org
both it works well. but from user at mydomain.org it reaches the callee one time on three.

my question is:
can I avoid NAPTR SRV rules and replace it by a TCP load balancing server ? (if I use full TCP for SIP messages and UDP as normal after media negotiation)
because it seems that NAPTR SRV work only if the caller proxy understands it. with a TCP load balancer I can avoid this,
but not sure if it won't break the SIP protocol.

Thanks

Franck
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From Mailings at kh-dev.de  Tue Mar 30 11:18:43 2010
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Tue, 30 Mar 2010 20:18:43 +0200
Subject: [Freeswitch-users] Snom wiki
Message-ID: <A2F1C0E212BBBA4E8C0E7BE568361F0403EDBFE52170@srv02.khdev.corp>

Just saw that the Snom interoperability page for FS was created/updated:
http://wiki.snom.com/Interoperability/PBX/FreeSwitch

Cheers, Klaus
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From neilp at cs.stanford.edu  Tue Mar 30 11:34:29 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 30 Mar 2010 11:34:29 -0700
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <6b65470d1003300927v7646585bh7c4e231dd7a86dcd@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com> 
	<e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com> 
	<6b65470d1003300926u3e85555ej942c84d021a7dbb7@mail.gmail.com> 
	<6b65470d1003300927v7646585bh7c4e231dd7a86dcd@mail.gmail.com>
Message-ID: <e5503bdb1003301134i794d0331w161518f05dcab7d@mail.gmail.com>

Old (Ubuntu) working FS install:
FreeSWITCH Version 1.0.trunk (15859)

New (CentOS) broken FS install:
FreeSWITCH Version 1.0.trunk (17071)

On Tue, Mar 30, 2010 at 9:27 AM, William Suffill
<william.suffill at gmail.com>wrote:

> Running fs_cli then running version also works.
> --
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.4 (exported)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 30 12:10:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 12:10:17 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <C7D78A78.7775%sean@obscuradigital.com>
References: <C7D78A78.7775%sean@obscuradigital.com>
Message-ID: <87f2f3b91003301210y55949fb5s69336088b532d696@mail.gmail.com>

The start_dtmf app simply turns on listening in-band for DTMFs, so this
should be just fine. Have bw.com enable in-band DTMFs and give it a test
drive.
-MC

On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hello,
>
> I?ve posted on the question on this topic before but I?ll try again with
> more detail.
>
> I have this svn version
> FreeSWITCH Version 1.0.trunk (16952)
> SIP provider: Bandwidth.com
>
> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
> recognize the incoming call tones.  Bandwidth can set up either inband or
> RC2833.  They require these codecs, G.711Ulaw and G.729A.
>
> The tech at Bandwidth wanted me to setup inband to test if that works
> better, but wasn?t sure where to set it up.  I added this to my dialplan
>
> <extension name="main_number_23" continue="true">
>       <condition field="destination_number" expression="^3000$">
>         <action application="answer"/>
>         <action application="start_dtmf"/>
>         <action application="sleep" data="1000"/>
>         <action application="conference" data="conference_1 at conference_1
> "/>
>       </condition>
> </extension>
>
> Should I add it to my sip trunk?
>
> Not sure how to troubleshoot this issue
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Tue Mar 30 12:37:27 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 12:37:27 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <87f2f3b91003301210y55949fb5s69336088b532d696@mail.gmail.com>
Message-ID: <C7D7A107.7790%sean@obscuradigital.com>

Yea still getting no love.  Had bandwidth turn on inband but still no
success.


On 3/30/10 12:10 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> The start_dtmf app simply turns on listening in-band for DTMFs, so this should
> be just fine. Have bw.com <http://bw.com>  enable in-band DTMFs and give it a
> test drive.
> -MC
> 
> On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello,
>> 
>> I?ve posted on the question on this topic before but I?ll try again with more
>> detail.
>> 
>> I have this svn version
>> FreeSWITCH Version 1.0.trunk (16952)
>> SIP provider: Bandwidth.com
>> 
>> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
>> recognize the incoming call tones. ?Bandwidth can set up either inband or
>> RC2833. ?They require these codecs, G.711Ulaw and G.729A.
>> 
>> The tech at Bandwidth wanted me to setup inband to test if that works better,
>> but wasn?t sure where to set it up. ?I added this to my dialplan
>> 
>> <extension name="main_number_23" continue="true">
>> ??????<condition field="destination_number" expression="^3000$">
>> ????????<action application="answer"/>
>> ????????<action application="start_dtmf"/>
>> ????????<action application="sleep" data="1000"/>
>> ????????<action application="conference" data="conference_1 at conference_1"/>
>> ??????</condition>
>> </extension>
>> 
>> Should I add it to my sip trunk?
>> 
>> Not sure how to troubleshoot this issue
>> 
>> Thanks
>> Sean 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lloydie.t at googlemail.com  Tue Mar 30 12:41:24 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Tue, 30 Mar 2010 20:41:24 +0100
Subject: [Freeswitch-users] Newbie MWI question
Message-ID: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>

I am more or less using the default config files with FS. I have a Mitel
5212 phone, which I leave a message for but the MWI light never illuminates
Any Ideas?
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From tculjaga at gmail.com  Tue Mar 30 12:42:03 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 30 Mar 2010 21:42:03 +0200
Subject: [Freeswitch-users] Radius/Authentication/Authorization
In-Reply-To: <COL109-W15576D4B6E4D7BFFFFEDD7F41F0@phx.gbl>
References: <COL109-W15576D4B6E4D7BFFFFEDD7F41F0@phx.gbl>
Message-ID: <65d96fc81003301242w425a6aedk5d6d0ad2c1638edf@mail.gmail.com>

hello,

here is an example



<configuration name="rad_auth.conf" description="radius authentification
module">
  <settings>
     <!-- backward compatibility to allow radiusclient config file instead
of an embedded config -->
     <!--  <param name="radius_config"
value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->
  </settings>

  <client>
    <param name="authserver" value="10.1.1.10:1812:gateway"/>
    <param name="dictionary"
value="/usr/local/etc/radiusclient/dictionary.all"/>
    <param name="seqfile" value="/var/run/radius.seq"/>
    <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>
    <param name="default_realm" value=""/>
    <param name="radius_timeout" value="3"/>
    <param name="radius_retries" value="2"/>
    <param name="radius_deadtime" value="0"/>
    <param name="bindaddr" value="*"/>
  </client>

  <vsas>
    <!--
         name:       just a description
         value:      direct input or variable
         pec:        vendor ID (0 for default, 9 for cisco...)
         expr:       1 for channel variable, 2 for direct input
         direction:  in for radius-request, out for radius-response
    -->

    <!-- mappings for radius request message; input attributes -->
    <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
direction="in"/>
    <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9"
expr="1" direction="in"/>
    <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1"
direction="in"/>
    <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0"
expr="1" direction="in"/>
    <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
direction="in"/>
    <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0"
direction="in"/>
    <param name="Login-User" id="1" value="1" pec="0" expr="0"
direction="in"/>

    <!-- mappings for radius-response message; output values from returning
outributes -->
    <param name="BILING_MODEL" id="109" value="billing_model" pec="9"
expr="0" direction="out"/>
    <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9"
expr="0" direction="out"/>
    <param name="CURRENCY" id="110" value="currency" pec="9" expr="0"
direction="out"/>
    <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9"
expr="0" direction="out"/>
    <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0"
direction="out"/>
    <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration"
pec="9" expr="0" direction="out"/>
    <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9"
expr="0" direction="out"/>
    <!-- expr param is to be ignored here-->
  </vsas>
</configuration>




in the dialplan you need to trigger auth as:

      <action application="auth_function" data="in ${DIALED_NUMBER}, in
${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>


there are two behaviours:

1. authorize the call according to username&pass and dialed number - if
authorized, the radius server returns credit time towards the dialed number
2. authorize the call according to username&pass - if authorized, the radius
server returns the current account balance






  <extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>

      <action inline="true" application="set"
data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set"
data="SERVICENUM=h323-prompt-id=${destination_number}"/>
      <action inline="true" application="set"
data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set"
data="CALLINGNUMBER=${caller_id_number}"/> -->
      <action inline="true" application="set"
data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>
<!--      <action inline="true" application="set" data="USERNAME=209354"/>
-->
      <action inline="true" application="set" data="PASSWD=003282"/>
<!--      <action inline="true" application="set"
data="DIALED_NUMBER=16094191500"/>  -->

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in
${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>


      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO
billing_model=${billing_model}"/>
      <action application="log" data="INFO
credit_amount=${credit_amount}"/>
      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO
preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>
      <action application="log" data="INFO
h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/>
-->
    </condition>
  </extension>


  <extension name="unitest_rad-ANI-balance">
    <condition field="destination_number" expression="^602$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>

      <action inline="true" application="set"
data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set"
data="SERVICENUM=h323-prompt-id=${destination_number}"/>
      <action inline="true" application="set"
data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
<!--      <action inline="true" application="set"
data="CALLINGNUMBER=${caller_id_number}"/> -->
      <action inline="true" application="set"
data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>
<!--      <action inline="true" application="set" data="USERNAME=209354"/>
-->
      <action inline="true" application="set" data="PASSWD=003282"/>
      <action inline="true" application="set"
data="DIALED_NUMBER=16094191500"/>

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in
${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>


      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
      <action application="log" data="INFO
billing_model=${billing_model}"/>
      <action application="log" data="INFO
credit_amount=${credit_amount}"/>
      <action application="log" data="INFO  currency=${currency}"/>
      <action application="log" data="INFO
preffered_lang=${preffered_lang}"/>
      <action application="log" data="INFO  credit_time=${credit_time}"/>
      <action application="log" data="INFO
h323_ivr_duration=${h323_ivr_duration}"/>
      <action application="log" data="INFO  return_code=${return_code}"/>
      <!-- <action application="execute_extension" data="AUTH XML default"/>
-->
    </condition>
  </extension>


will update the wiki by the end of the week.


you have enough information for now.

Tihomir.



On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch at live.com>wrote:

>  Hi Neal and other Contributors to FS,
>
> I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS
> to download and install it by "make current" and it is successfully built
> and installed. Could you please outline the detailed steps to do all this
> configuration.
>
> If it is still not ready, is there any other method that somebody has
> already implemented like perl scripts etc. Any help in this regard is badly
> required. THanks for your cooperation.
>
> Regards
> ----------
> Abid Saleem
> Sr. Product Manager
>
>
> ------------------------------
> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Tue Mar 30 12:50:12 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 30 Mar 2010 15:50:12 -0400
Subject: [Freeswitch-users] DTMF
In-Reply-To: <C7D7A107.7790%sean@obscuradigital.com>
References: <87f2f3b91003301210y55949fb5s69336088b532d696@mail.gmail.com>
	<C7D7A107.7790%sean@obscuradigital.com>
Message-ID: <367751821003301250k4152c372n53f601ac29258ecf@mail.gmail.com>

Use wireshark to see (and playback) the DTMF coming down the line.

For reliable inband I believe you need to be using 711 - so that is worth
checking too.

On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Yea still getting no love.  Had bandwidth turn on inband but still no
> success.
>
>
> On 3/30/10 12:10 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> The start_dtmf app simply turns on listening in-band for DTMFs, so this
> should be just fine. Have bw.com <http://bw.com>  enable in-band DTMFs and
> give it a test drive.
>
> -MC
>
> On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Hello,
>
> I?ve posted on the question on this topic before but I?ll try again with
> more detail.
>
> I have this svn version
> FreeSWITCH Version 1.0.trunk (16952)
> SIP provider: Bandwidth.com
>
> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
> recognize the incoming call tones.  Bandwidth can set up either inband or
> RC2833.  They require these codecs, G.711Ulaw and G.729A.
>
> The tech at Bandwidth wanted me to setup inband to test if that works
> better, but wasn?t sure where to set it up.  I added this to my dialplan
>
> <extension name="main_number_23" continue="true">
>       <condition field="destination_number" expression="^3000$">
>         <action application="answer"/>
>         <action application="start_dtmf"/>
>         <action application="sleep" data="1000"/>
>         <action application="conference" data="conference_1 at conference_1
> "/>
>       </condition>
> </extension>
>
> Should I add it to my sip trunk?
>
> Not sure how to troubleshoot this issue
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oseslija at gmail.com  Tue Mar 30 12:53:39 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 30 Mar 2010 21:53:39 +0200
Subject: [Freeswitch-users] domain name in shared DB
In-Reply-To: <9099D7E46BBB410B893CF5BEC23C3C26@MOBILEE1705>
References: <9099D7E46BBB410B893CF5BEC23C3C26@MOBILEE1705>
Message-ID: <4468a6771003301253k4646baceo9d016d8b6c5d7b73@mail.gmail.com>

People use outbound proxy for that purpose, so all traffic goes to it.

O.

On Tue, Mar 30, 2010 at 7:18 PM, Madovsky <infos at madovsky.org> wrote:

>  2 nodes in shared DB, node1 and node2 respectively with hostname
> node1.mydomain.org node2.mydomain.org
> the profile name is called mydomain.org which is shared by the 2 nodes.
> a user registers on node1 and wait calls.
> someone from external gateway call him, so user at node1.mydomain.org or
> user at node2.mydomain.org
> both it works well. but from user at mydomain.org it reaches the callee one
> time on three.
>
> my question is:
> can I avoid NAPTR SRV rules and replace it by a TCP load balancing server ?
> (if I use full TCP for SIP messages and UDP as normal after media
> negotiation)
> because it seems that NAPTR SRV work only if the caller proxy understands
> it. with a TCP load balancer I can avoid this,
> but not sure if it won't break the SIP protocol.
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue Mar 30 13:08:32 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 30 Mar 2010 13:08:32 -0700
Subject: [Freeswitch-users] Cisco 7940 To Private
In-Reply-To: <E151C0A3-250B-43BB-A399-C54D8B71137D@freeswitch.org>
References: <51FE9B78-4CBE-4352-B250-B76F4CDD8B3B@tlainvestments.com>
	<E151C0A3-250B-43BB-A399-C54D8B71137D@freeswitch.org>
Message-ID: <72C1BE0B-5008-4D53-9298-89C26AC71377@tlainvestments.com>

Thanks!  I'll give it a try - I'm a couple weeks behind.


On Mar 30, 2010, at 10:53 AM, Brian West wrote:

> svn update this is fixed already.
> 
> /b
> 
> On Mar 30, 2010, at 12:41 PM, Troy Anderson wrote:
> 
>> I have some Cisco 7940's registered and working well with FS, but whenever I dial out from the Cisco phone, it indicates the call is "To Private" rather than showing the number I'm calling or something.  I've put the SIP trace in pastebin http://pastebin.freeswitch.org/12589 - there is a Remote-Part-ID field that seems OK.  Does anyone know what Cisco is looking for here?
>> 
>> Thanks,
>> Troy
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue Mar 30 13:12:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Mar 2010 15:12:26 -0500
Subject: [Freeswitch-users] Cisco 7940 To Private
In-Reply-To: <72C1BE0B-5008-4D53-9298-89C26AC71377@tlainvestments.com>
References: <51FE9B78-4CBE-4352-B250-B76F4CDD8B3B@tlainvestments.com>
	<E151C0A3-250B-43BB-A399-C54D8B71137D@freeswitch.org>
	<72C1BE0B-5008-4D53-9298-89C26AC71377@tlainvestments.com>
Message-ID: <C6D99321-8681-47A2-80CF-E9D30BACD9F6@freeswitch.org>

it was fixed last thursday. :P

/b

On Mar 30, 2010, at 3:08 PM, Troy Anderson wrote:

> Thanks!  I'll give it a try - I'm a couple weeks behind.
> 
> 
> On Mar 30, 2010, at 10:53 AM, Brian West wrote:
> 
>> svn update this is fixed already.
>> 
>> /b



From fraserredmond at gmail.com  Tue Mar 30 13:19:03 2010
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Tue, 30 Mar 2010 21:19:03 +0100
Subject: [Freeswitch-users] Favorite SIP Providers? Australian providers?
Message-ID: <c32ee1cf1003301319y4cb38049nf9e9dffeb217147f@mail.gmail.com>

I was looking through the list of Providers with example configs in the
wiki:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

I was wondering if anyone had a provider that they really loved? And why?

I've got inbound 1800/0800 in UK & USA already, but might change. Also
interested in outbound.

Also, I want to get a 1800 number in Australia - any suggestions?


Sorry in advance if this is one of those questions that does the rounds a
couple of times a year (I did have a look through the archives via Google
and didn't see anything), or if I'm treading on dangerous ground asking for
recommendations of commercial services - so I guess only recommend services
you use, not services you provide ;-)

Cheers,
Fraser
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From msc at freeswitch.org  Tue Mar 30 13:45:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 13:45:33 -0700
Subject: [Freeswitch-users] Newbie MWI question
In-Reply-To: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
References: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
Message-ID: <v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>

Is that a SIP phone?

On Tue, Mar 30, 2010 at 12:41 PM, lloyd thomas <lloydie.t at googlemail.com>wrote:

> I am more or less using the default config files with FS. I have a Mitel
> 5212 phone, which I leave a message for but the MWI light never illuminates
> Any Ideas?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Tue Mar 30 14:09:34 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 30 Mar 2010 23:09:34 +0200
Subject: [Freeswitch-users] Newbie MWI question
In-Reply-To: <v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>
References: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
	<v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>
Message-ID: <6358F826-7276-440A-A8D2-7D0AB6F75A6A@gmail.com>

Michael,

Yes, as many Mitel phones, I think they can do both MiNet protocol and  
SIP.
I dunno how good is their SIP though...

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/03/2010 ? 22:45, Michael Collins a ?crit :

> Is that a SIP phone?
>
> On Tue, Mar 30, 2010 at 12:41 PM, lloyd thomas <lloydie.t at googlemail.com 
> > wrote:
> I am more or less using the default config files with FS. I have a  
> Mitel 5212 phone, which I leave a message for but the MWI light  
> never illuminates
> Any Ideas?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jerry.richards at teotech.com  Tue Mar 30 14:23:13 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 30 Mar 2010 14:23:13 -0700
Subject: [Freeswitch-users] Calling Phone Behind Netgear Router (with NAT)
Message-ID: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>

We are trying call a phone that resides behind a Netgear router (with NAT).
The Bria softphone registers and we can make calls from the softphone, but
we cannot make calls to the softphone.

                WAN
------------  public IP -------- 10.10.10.xx ----------------
|Freeswitch|<---------->|Router|<----------->|Bria softphone|
------------            --------             ----------------

Does anyone know of good documentation/example showing this configuration?

Best Regards,
Jerry



From kenfulmer at icstechnologysolutions.com  Tue Mar 30 14:30:04 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 30 Mar 2010 16:30:04 -0500
Subject: [Freeswitch-users] Bridge two calls
Message-ID: <018c01cad050$29f9fa90$7dedefb0$@com>

I need to bridge two calls together: one from an ITSP and one from an
internal PBX. I've tried the following syntax but can't get it to work. Can
someone clarify what I'm doing wrong?

 

Here are the two gateway configs located in the Sip Profiles / External
directory:

 

<include>

    <gateway name="IPCM-VLAB-1">

      <param name="proxy" value="10.10.3.10"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="SDC1-VGW-1">

      <param name="proxy" value="172.16.15.11"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

Here is the inbound route in the Dial Plan / Public directory (I'm just
trying to get one number working at the moment):

 

<extension name="From SDC1-VGW-1">

   <condition field="destination_number" expression="^(2053148491)$">

       <action application="bridge" data="sofia/external/IPCM-VLAB-1"/>

   </condition>

</extension>

 

I've also tried the gateway statement with no luck either:

 

  <action application="bridge" data="sofia/gateway/mygatewayname/555123456"
/>

 

Any ideas?

 

Thanks,

 

Ken Fulmer

 

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From brian at freeswitch.org  Tue Mar 30 14:33:16 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 30 Mar 2010 16:33:16 -0500
Subject: [Freeswitch-users] Calling Phone Behind Netgear Router (with
	NAT)
In-Reply-To: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>
References: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>
Message-ID: <983716D7-6E42-4341-9DB9-473C6F493EC8@freeswitch.org>

Its called a SIP ALG and those netgears are busted.  See if it has an option to turn it off.

/b

On Mar 30, 2010, at 4:23 PM, Jerry Richards wrote:

> Does anyone know of good documentation/example showing this configuration?



From msc at freeswitch.org  Tue Mar 30 14:38:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 14:38:40 -0700
Subject: [Freeswitch-users] Bridge two calls
In-Reply-To: <018c01cad050$29f9fa90$7dedefb0$@com>
References: <018c01cad050$29f9fa90$7dedefb0$@com>
Message-ID: <r2y87f2f3b91003301438z1a850606q5ae94830d5630d89@mail.gmail.com>

You're getting there. You were correct to use the public.xml file. However,
your syntax wasn't correct:

<!-- Oops -->
<action application="bridge" data="sofia/external/IPCM-VLAB-1"/>

<!-- Correct -->
<action application="bridge" data="sofia/gateway/IPCM-VLAB-1/$1"/>

Question: what about calls from the internal PBX to the ITSP? Do you need to
route calls that way as well?
-MC

On Tue, Mar 30, 2010 at 2:30 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  I need to bridge two calls together: one from an ITSP and one from an
> internal PBX. I?ve tried the following syntax but can?t get it to work. Can
> someone clarify what I?m doing wrong?
>
>
>
> Here are the two gateway configs located in the Sip Profiles / External
> directory:
>
>
>
> <include>
>
>     <gateway name="IPCM-VLAB-1">
>
>       <param name="proxy" value="10.10.3.10"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> <include>
>
>     <gateway name="SDC1-VGW-1">
>
>       <param name="proxy" value="172.16.15.11"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> Here is the inbound route in the Dial Plan / Public directory (I?m just
> trying to get one number working at the moment):
>
>
>
> <extension name="From SDC1-VGW-1">
>
>    <condition field="destination_number" expression="^(2053148491)$">
>
>        <action application="bridge" data="sofia/external/IPCM-VLAB-1"/>
>
>    </condition>
>
> </extension>
>
>
>
> I?ve also tried the gateway statement with no luck either:
>
>
>
>   <action application="bridge" data="sofia/gateway/mygatewayname/555123456"
> />
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 30 14:39:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 14:39:43 -0700
Subject: [Freeswitch-users] Newbie MWI question
In-Reply-To: <6358F826-7276-440A-A8D2-7D0AB6F75A6A@gmail.com>
References: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
	<v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>
	<6358F826-7276-440A-A8D2-7D0AB6F75A6A@gmail.com>
Message-ID: <x2g87f2f3b91003301439ya29f1bc7y4207825ddf8f63d8@mail.gmail.com>

2010/3/30 David Ponzone <david.ponzone at gmail.com>

> Michael,
>
> Yes, as many Mitel phones, I think they can do both MiNet protocol and SIP.
> I dunno how good is their SIP though...
>

Well, FreeSWITCH ne parles pas MiNet! :) You'll need to try SIP and hope for
the best.
-MC


>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 30/03/2010 ? 22:45, Michael Collins a ?crit :
>
> Is that a SIP phone?
>
> On Tue, Mar 30, 2010 at 12:41 PM, lloyd thomas <lloydie.t at googlemail.com>wrote:
>
>> I am more or less using the default config files with FS. I have a Mitel
>> 5212 phone, which I leave a message for but the MWI light never illuminates
>> Any Ideas?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Tue Mar 30 14:40:10 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 14:40:10 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <367751821003301250k4152c372n53f601ac29258ecf@mail.gmail.com>
Message-ID: <C7D7BDCA.77A5%sean@obscuradigital.com>

Ok I?ve run wireshark, but not sure what to look for.  It?s using codec 711
in the rtp stream, but not sure what to look for in regards to DTMF?

Thanks
Sean


On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa at gmail.com> wrote:

> Use wireshark to see (and playback) the DTMF coming down the line.
> 
> For reliable inband I believe you need to be using 711 - so that is worth
> checking too.
> 
> On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Yea still getting no love. ?Had bandwidth turn on inband but still no
>> success.
>> 
>> 
>> On 3/30/10 12:10 PM, "Michael Collins" <msc at freeswitch.org
>> <http://msc at freeswitch.org> > wrote:
>> 
>>> The start_dtmf app simply turns on listening in-band for DTMFs, so this
>>> should be just fine. Have bw.com <http://bw.com>  <http://bw.com> ?enable
>>> in-band DTMFs and give it a test drive.
>>> 
>>> -MC
>>> 
>>> On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com
>>> <http://sean at obscuradigital.com> > wrote:
>>>> Hello,
>>>> 
>>>> I?ve posted on the question on this topic before but I?ll try again with
>>>> more detail.
>>>> 
>>>> I have this svn version
>>>> FreeSWITCH Version 1.0.trunk (16952)
>>>> SIP provider: Bandwidth.com
>>>> 
>>>> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
>>>> recognize the incoming call tones. ?Bandwidth can set up either inband or
>>>> RC2833. ?They require these codecs, G.711Ulaw and G.729A.
>>>> 
>>>> The tech at Bandwidth wanted me to setup inband to test if that works
>>>> better, but wasn?t sure where to set it up. ?I added this to my dialplan
>>>> 
>>>> <extension name="main_number_23" continue="true">
>>>> ??????<condition field="destination_number" expression="^3000$">
>>>> ????????<action application="answer"/>
>>>> ????????<action application="start_dtmf"/>
>>>> ????????<action application="sleep" data="1000"/>
>>>> ????????<action application="conference" data="conference_1 at conference_1"/>
>>>> ??????</condition>
>>>> </extension>
>>>> 
>>>> Should I add it to my sip trunk?
>>>> 
>>>> Not sure how to troubleshoot this issue
>>>> 
>>>> Thanks
>>>> Sean 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Mar 30 14:44:13 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 14:44:13 -0700
Subject: [Freeswitch-users] Calling Phone Behind Netgear Router (with
	NAT)
In-Reply-To: <983716D7-6E42-4341-9DB9-473C6F493EC8@freeswitch.org>
References: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>
	<983716D7-6E42-4341-9DB9-473C6F493EC8@freeswitch.org>
Message-ID: <r2q87f2f3b91003301444xf5ea9cffub50787b933c7e1da@mail.gmail.com>

On Tue, Mar 30, 2010 at 2:33 PM, Brian West <brian at freeswitch.org> wrote:

> Its called a SIP ALG and those netgears are busted.  See if it has an
> option to turn it off.
>
> /b
>
> I successfully turned off the SIP ALG in a NetGear using this procedure:

Unplug power from unit
Place unit underneath front driver-side tire of your car
Start car, put in drive, remove foot from brake
As car moves forward, hear crunching noise
When crunching noise ends apply brake and place car in reverse
Remove foot from brake
As car moves backward, hear more crunching sounds
Repeat as needed
When crunching sounds have subsided the SIP ALG is successfully deactivated.

-MC
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From msc at freeswitch.org  Tue Mar 30 14:46:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 14:46:02 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <C7D7BDCA.77A5%sean@obscuradigital.com>
References: <367751821003301250k4152c372n53f601ac29258ecf@mail.gmail.com>
	<C7D7BDCA.77A5%sean@obscuradigital.com>
Message-ID: <y2v87f2f3b91003301446z898ca789h3d310f13010e3e8c@mail.gmail.com>

Wireshark lets you play back the audio of the call. It also shows any
RFC2833 DTMF sequences. I'd start by listening to the audio. Do you hear any
DTMFs?
-MC

On Tue, Mar 30, 2010 at 2:40 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Ok I?ve run wireshark, but not sure what to look for.  It?s using codec
> 711 in the rtp stream, but not sure what to look for in regards to DTMF?
>
> Thanks
> Sean
>
>
> On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa at gmail.com> wrote:
>
> Use wireshark to see (and playback) the DTMF coming down the line.
>
> For reliable inband I believe you need to be using 711 - so that is worth
> checking too.
>
> On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Yea still getting no love.  Had bandwidth turn on inband but still no
> success.
>
>
> On 3/30/10 12:10 PM, "Michael Collins" <msc at freeswitch.org <
> http://msc at freeswitch.org> > wrote:
>
> The start_dtmf app simply turns on listening in-band for DTMFs, so this
> should be just fine. Have bw.com <http://bw.com>  <http://bw.com>  enable
> in-band DTMFs and give it a test drive.
>
> -MC
>
> On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com <
> http://sean at obscuradigital.com> > wrote:
>
> Hello,
>
> I?ve posted on the question on this topic before but I?ll try again with
> more detail.
>
> I have this svn version
> FreeSWITCH Version 1.0.trunk (16952)
> SIP provider: Bandwidth.com
>
> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
> recognize the incoming call tones.  Bandwidth can set up either inband or
> RC2833.  They require these codecs, G.711Ulaw and G.729A.
>
> The tech at Bandwidth wanted me to setup inband to test if that works
> better, but wasn?t sure where to set it up.  I added this to my dialplan
>
> <extension name="main_number_23" continue="true">
>       <condition field="destination_number" expression="^3000$">
>         <action application="answer"/>
>         <action application="start_dtmf"/>
>         <action application="sleep" data="1000"/>
>         <action application="conference" data="conference_1 at conference_1
> "/>
>       </condition>
> </extension>
>
> Should I add it to my sip trunk?
>
> Not sure how to troubleshoot this issue
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloydie.t at googlemail.com  Tue Mar 30 14:52:01 2010
From: lloydie.t at googlemail.com (lloyd thomas)
Date: Tue, 30 Mar 2010 22:52:01 +0100
Subject: [Freeswitch-users] Newbie MWI question
In-Reply-To: <x2g87f2f3b91003301439ya29f1bc7y4207825ddf8f63d8@mail.gmail.com>
References: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
	<v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>
	<6358F826-7276-440A-A8D2-7D0AB6F75A6A@gmail.com>
	<x2g87f2f3b91003301439ya29f1bc7y4207825ddf8f63d8@mail.gmail.com>
Message-ID: <8b61bd671003301452t3f58a56cjfdb818eae2ae30@mail.gmail.com>

Apologies. There was a setting I missed in the config. Unfortunately the
setting for the voicemail number is in a different place/page for the
setting of the voicemail server address (which I over looked).
Any way Mitel in SIP mode seems to work fine so far!


Lloydie T

On 30 March 2010 22:39, Michael Collins <msc at freeswitch.org> wrote:

>
>
> 2010/3/30 David Ponzone <david.ponzone at gmail.com>
>
> Michael,
>>
>> Yes, as many Mitel phones, I think they can do both MiNet protocol and
>> SIP.
>> I dunno how good is their SIP though...
>>
>
> Well, FreeSWITCH ne parles pas MiNet! :) You'll need to try SIP and hope
> for the best.
> -MC
>
>
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 30/03/2010 ? 22:45, Michael Collins a ?crit :
>>
>> Is that a SIP phone?
>>
>> On Tue, Mar 30, 2010 at 12:41 PM, lloyd thomas <lloydie.t at googlemail.com>wrote:
>>
>>> I am more or less using the default config files with FS. I have a Mitel
>>> 5212 phone, which I leave a message for but the MWI light never illuminates
>>> Any Ideas?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kenfulmer at icstechnologysolutions.com  Tue Mar 30 14:57:44 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 30 Mar 2010 16:57:44 -0500
Subject: [Freeswitch-users] Bridge two calls
In-Reply-To: <r2y87f2f3b91003301438z1a850606q5ae94830d5630d89@mail.gmail.com>
References: <018c01cad050$29f9fa90$7dedefb0$@com>
	<r2y87f2f3b91003301438z1a850606q5ae94830d5630d89@mail.gmail.com>
Message-ID: <01bb01cad054$076c5190$1644f4b0$@com>

Yes, we do. I assume I can just reverse the statements to send calls in the
other direction?

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, March 30, 2010 4:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Bridge two calls

 

You're getting there. You were correct to use the public.xml file. However,
your syntax wasn't correct:

<!-- Oops -->
<action application="bridge" data="sofia/external/IPCM-VLAB-1"/>

<!-- Correct -->
<action application="bridge" data="sofia/gateway/IPCM-VLAB-1/$1"/>

Question: what about calls from the internal PBX to the ITSP? Do you need to
route calls that way as well?
-MC

On Tue, Mar 30, 2010 at 2:30 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

I need to bridge two calls together: one from an ITSP and one from an
internal PBX. I've tried the following syntax but can't get it to work. Can
someone clarify what I'm doing wrong?

 

Here are the two gateway configs located in the Sip Profiles / External
directory:

 

<include>

    <gateway name="IPCM-VLAB-1">

      <param name="proxy" value="10.10.3.10"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

<include>

    <gateway name="SDC1-VGW-1">

      <param name="proxy" value="172.16.15.11"/>

      <param name="register" value="false"/>

      <param name="context" value="public"/>

    </gateway>

</include>

 

Here is the inbound route in the Dial Plan / Public directory (I'm just
trying to get one number working at the moment):

 

<extension name="From SDC1-VGW-1">

   <condition field="destination_number" expression="^(2053148491)$">

       <action application="bridge" data="sofia/external/IPCM-VLAB-1"/>

   </condition>

</extension>

 

I've also tried the gateway statement with no luck either:

 

  <action application="bridge" data="sofia/gateway/mygatewayname/555123456"
/>

 

Any ideas?

 

Thanks,

 

Ken Fulmer

 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From msc at freeswitch.org  Tue Mar 30 15:03:55 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 15:03:55 -0700
Subject: [Freeswitch-users] Bridge two calls
In-Reply-To: <01bb01cad054$076c5190$1644f4b0$@com>
References: <018c01cad050$29f9fa90$7dedefb0$@com>
	<r2y87f2f3b91003301438z1a850606q5ae94830d5630d89@mail.gmail.com>
	<01bb01cad054$076c5190$1644f4b0$@com>
Message-ID: <s2z87f2f3b91003301503nec72137du9e751ee1895e8c11@mail.gmail.com>

On Tue, Mar 30, 2010 at 2:57 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Yes, we do. I assume I can just reverse the statements to send calls in
> the other direction?
>
Mostly yes. However you need to handle a range of phone numbers in your
regular expression instead of a specific 10-digit phone number. The Dialplan
XML page on the wiki has quite a few examples...

-MC

>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, March 30, 2010 4:39 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Bridge two calls
>
>
>
> You're getting there. You were correct to use the public.xml file. However,
> your syntax wasn't correct:
>
> <!-- Oops -->
> <action application="bridge" data="sofia/external/IPCM-VLAB-1"/>
>
> <!-- Correct -->
> <action application="bridge" data="sofia/gateway/IPCM-VLAB-1/$1"/>
>
> Question: what about calls from the internal PBX to the ITSP? Do you need
> to route calls that way as well?
> -MC
>
> On Tue, Mar 30, 2010 at 2:30 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> I need to bridge two calls together: one from an ITSP and one from an
> internal PBX. I?ve tried the following syntax but can?t get it to work. Can
> someone clarify what I?m doing wrong?
>
>
>
> Here are the two gateway configs located in the Sip Profiles / External
> directory:
>
>
>
> <include>
>
>     <gateway name="IPCM-VLAB-1">
>
>       <param name="proxy" value="10.10.3.10"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> <include>
>
>     <gateway name="SDC1-VGW-1">
>
>       <param name="proxy" value="172.16.15.11"/>
>
>       <param name="register" value="false"/>
>
>       <param name="context" value="public"/>
>
>     </gateway>
>
> </include>
>
>
>
> Here is the inbound route in the Dial Plan / Public directory (I?m just
> trying to get one number working at the moment):
>
>
>
> <extension name="From SDC1-VGW-1">
>
>    <condition field="destination_number" expression="^(2053148491)$">
>
>        <action application="bridge" data="sofia/external/IPCM-VLAB-1"/>
>
>    </condition>
>
> </extension>
>
>
>
> I?ve also tried the gateway statement with no luck either:
>
>
>
>   <action application="bridge" data="sofia/gateway/mygatewayname/555123456"
> />
>
>
>
> Any ideas?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Mar 30 15:33:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 15:33:35 -0700
Subject: [Freeswitch-users] Favorite SIP Providers? Australian providers?
In-Reply-To: <c32ee1cf1003301319y4cb38049nf9e9dffeb217147f@mail.gmail.com>
References: <c32ee1cf1003301319y4cb38049nf9e9dffeb217147f@mail.gmail.com>
Message-ID: <g2p87f2f3b91003301533n44bef6e0w5325c6559f55dfb2@mail.gmail.com>

On Tue, Mar 30, 2010 at 1:19 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> I was looking through the list of Providers with example configs in the
> wiki:
> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
>
> I was wondering if anyone had a provider that they really loved? And why?
>
> I've got inbound 1800/0800 in UK & USA already, but might change. Also
> interested in outbound.
>
> Also, I want to get a 1800 number in Australia - any suggestions?
>

We have a provider in Australia about whom we've heard many good things:
netsip.com.au
Go check them out. They have an entry in the FS wiki under provider configs
as well.

-MC
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From jaybinks at gmail.com  Tue Mar 30 15:51:00 2010
From: jaybinks at gmail.com (jay binks)
Date: Wed, 31 Mar 2010 08:51:00 +1000
Subject: [Freeswitch-users] Favorite SIP Providers? Australian providers?
In-Reply-To: <g2p87f2f3b91003301533n44bef6e0w5325c6559f55dfb2@mail.gmail.com>
References: <c32ee1cf1003301319y4cb38049nf9e9dffeb217147f@mail.gmail.com> 
	<g2p87f2f3b91003301533n44bef6e0w5325c6559f55dfb2@mail.gmail.com>
Message-ID: <ec2ce17e1003301551i3b57d808j988654d24e0d9920@mail.gmail.com>

> We have a provider in Australia about whom we've heard many good things:
> netsip.com.au

+1 from me  :)

these guys will look after you for sure,
and they support the Freeswitch project wherever possible.

-- 
Sincerely

Jay


From sean at obscuradigital.com  Tue Mar 30 15:56:52 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 15:56:52 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <367751821003301250k4152c372n53f601ac29258ecf@mail.gmail.com>
Message-ID: <C7D7CFC4.77BE%sean@obscuradigital.com>

Unfortunately I don?t see any reference to DTMF coming from outside source.

It shows it?s using 711

Thanks
Sean


On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa at gmail.com> wrote:

> Use wireshark to see (and playback) the DTMF coming down the line.
> 
> For reliable inband I believe you need to be using 711 - so that is worth
> checking too.
> 
> On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Yea still getting no love. ?Had bandwidth turn on inband but still no
>> success.
>> 
>> 
>> On 3/30/10 12:10 PM, "Michael Collins" <msc at freeswitch.org
>> <http://msc at freeswitch.org> > wrote:
>> 
>>> The start_dtmf app simply turns on listening in-band for DTMFs, so this
>>> should be just fine. Have bw.com <http://bw.com>  <http://bw.com> ?enable
>>> in-band DTMFs and give it a test drive.
>>> 
>>> -MC
>>> 
>>> On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean at obscuradigital.com
>>> <http://sean at obscuradigital.com> > wrote:
>>>> Hello,
>>>> 
>>>> I?ve posted on the question on this topic before but I?ll try again with
>>>> more detail.
>>>> 
>>>> I have this svn version
>>>> FreeSWITCH Version 1.0.trunk (16952)
>>>> SIP provider: Bandwidth.com
>>>> 
>>>> I?m assuming that DTMF rc2833 is the default, but I can?t get my ivr to
>>>> recognize the incoming call tones. ?Bandwidth can set up either inband or
>>>> RC2833. ?They require these codecs, G.711Ulaw and G.729A.
>>>> 
>>>> The tech at Bandwidth wanted me to setup inband to test if that works
>>>> better, but wasn?t sure where to set it up. ?I added this to my dialplan
>>>> 
>>>> <extension name="main_number_23" continue="true">
>>>> ??????<condition field="destination_number" expression="^3000$">
>>>> ????????<action application="answer"/>
>>>> ????????<action application="start_dtmf"/>
>>>> ????????<action application="sleep" data="1000"/>
>>>> ????????<action application="conference" data="conference_1 at conference_1"/>
>>>> ??????</condition>
>>>> </extension>
>>>> 
>>>> Should I add it to my sip trunk?
>>>> 
>>>> Not sure how to troubleshoot this issue
>>>> 
>>>> Thanks
>>>> Sean 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Mar 30 16:11:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 16:11:22 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <C7D7CFC4.77BE%sean@obscuradigital.com>
References: <367751821003301250k4152c372n53f601ac29258ecf@mail.gmail.com>
	<C7D7CFC4.77BE%sean@obscuradigital.com>
Message-ID: <m2z87f2f3b91003301611md91a341fq3bcac8c66c2d129b@mail.gmail.com>

On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Unfortunately I don?t see any reference to DTMF coming from outside
> source.
>
> It shows it?s using 711
>
> Thanks
> Sean
>

What about listening to the call? Do you hear any DTMFs in the audio stream?
If not then it looks like the far end isn't sending DTMFs which means you
will need to talk to your provider to continue troubleshooting.

-MC
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From sean at obscuradigital.com  Tue Mar 30 16:29:21 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 16:29:21 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <m2z87f2f3b91003301611md91a341fq3bcac8c66c2d129b@mail.gmail.com>
Message-ID: <C7D7D761.77C8%sean@obscuradigital.com>

Thanks man...

Looking closer at the wireshark capture I?m only seeing RTP going out and
nothing coming.

I?ll contact Bandwidth again

Sean


On 3/30/10 4:11 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 
> 
> On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Unfortunately I don?t see any reference to DTMF coming from outside source.
>> 
>> It shows it?s using 711
>> 
>> Thanks
>> Sean
> 
> What about listening to the call? Do you hear any DTMFs in the audio stream?
> If not then it looks like the far end isn't sending DTMFs which means you will
> need to talk to your provider to continue troubleshooting.
> 
> -MC
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sean at obscuradigital.com  Tue Mar 30 17:09:36 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 17:09:36 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <m2z87f2f3b91003301611md91a341fq3bcac8c66c2d129b@mail.gmail.com>
Message-ID: <C7D7E0D0.77CA%sean@obscuradigital.com>

So it appears that Bandwidth requires the audio to be routed on the external
public IP address, but since my freeswitch pbx box is behind a NAT the audio
is trying to route to a private IP.  The tech at bandwidth says that won?t
work.

Not sure how to correct this?

Thanks
Sean


On 3/30/10 4:11 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 
> 
> On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Unfortunately I don?t see any reference to DTMF coming from outside source.
>> 
>> It shows it?s using 711
>> 
>> Thanks
>> Sean
> 
> What about listening to the call? Do you hear any DTMFs in the audio stream?
> If not then it looks like the far end isn't sending DTMFs which means you will
> need to talk to your provider to continue troubleshooting.
> 
> -MC
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Mar 30 17:29:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Mar 2010 17:29:16 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <C7D7E0D0.77CA%sean@obscuradigital.com>
References: <m2z87f2f3b91003301611md91a341fq3bcac8c66c2d129b@mail.gmail.com>
	<C7D7E0D0.77CA%sean@obscuradigital.com>
Message-ID: <w2p87f2f3b91003301729ic73dd418u8a968668499d79db@mail.gmail.com>

On Tue, Mar 30, 2010 at 5:09 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  So it appears that Bandwidth requires the audio to be routed on the
> external public IP address, but since my freeswitch pbx box is behind a NAT
> the audio is trying to route to a private IP.  The tech at bandwidth says
> that won?t work.
>
> Not sure how to correct this?
>
> Thanks
> Sean
>
> I strongly recommend you read up on NAT:
http://wiki.freeswitch.org/wiki/Nat

and

http://wiki.freeswitch.org/wiki/NAT_Traversal

and

http://wiki.freeswitch.org/wiki/Auto_NAT

You need to set up your external profile properly, which means you need to
understand your network and your NAT device. Did you turn off any SIP ALG's
in your device? Do you have a static public IP address? Those are important
questions to answer before you continue.

-MC


>
> On 3/30/10 4:11 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>
>
> On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean at obscuradigital.com>
> wrote:
>
> Unfortunately I don?t see any reference to DTMF coming from outside source.
>
> It shows it?s using 711
>
> Thanks
> Sean
>
>
> What about listening to the call? Do you hear any DTMFs in the audio
> stream? If not then it looks like the far end isn't sending DTMFs which
> means you will need to talk to your provider to continue troubleshooting.
>
> -MC
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Tue Mar 30 18:46:04 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 30 Mar 2010 18:46:04 -0700
Subject: [Freeswitch-users] DTMF
In-Reply-To: <w2p87f2f3b91003301729ic73dd418u8a968668499d79db@mail.gmail.com>
Message-ID: <C7D7F76C.77D3%sean@obscuradigital.com>

Thanks michael for all the help.

I?ve seem to have fixed this by adding the public IP address in my sip
profile file under the ext-rtp-ip and ext-sip-ip param.  Now I can see the
DTMF tones come in while capturing.

Sean


On 3/30/10 5:29 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 
> 
> On Tue, Mar 30, 2010 at 5:09 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> So it appears that Bandwidth requires the audio to be routed on the external
>> public IP address, but since my freeswitch pbx box is behind a NAT the audio
>> is trying to route to a private IP. ?The tech at bandwidth says that won?t
>> work.
>> 
>> Not sure how to correct this?
>> 
>> Thanks
>> Sean
>> 
> I strongly recommend you read up on NAT:
> http://wiki.freeswitch.org/wiki/Nat
> 
> and
> 
> http://wiki.freeswitch.org/wiki/NAT_Traversal
> 
> and
> 
> http://wiki.freeswitch.org/wiki/Auto_NAT
> 
> You need to set up your external profile properly, which means you need to
> understand your network and your NAT device. Did you turn off any SIP ALG's in
> your device? Do you have a static public IP address? Those are important
> questions to answer before you continue.
> 
> -MC
>  
>> 
>> 
>> On 3/30/10 4:11 PM, "Michael Collins" <msc at freeswitch.org
>> <http://msc at freeswitch.org> > wrote:
>> 
>>> 
>>> 
>>> On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean at obscuradigital.com
>>> <http://sean at obscuradigital.com> > wrote:
>>>> Unfortunately I don?t see any reference to DTMF coming from outside source.
>>>> 
>>>> It shows it?s using 711
>>>> 
>>>> Thanks
>>>> Sean
>>> 
>>> What about listening to the call? Do you hear any DTMFs in the audio stream?
>>> If not then it looks like the far end isn't sending DTMFs which means you
>>> will need to talk to your provider to continue troubleshooting.
>>> 
>>> -MC
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From elihayun at gmail.com  Tue Mar 30 21:56:54 2010
From: elihayun at gmail.com (Eli Hayun)
Date: Wed, 31 Mar 2010 07:56:54 +0300
Subject: [Freeswitch-users] Variable "default_language" has no effect
 on	javascript
In-Reply-To: <87f2f3b91003250942h77de93a6y4f8b9ee1767b7fd9@mail.gmail.com>
References: <4BAB02CC.8040602@gmail.com>
	<87f2f3b91003250942h77de93a6y4f8b9ee1767b7fd9@mail.gmail.com>
Message-ID: <4BB2D616.1080109@savion.huji.ac.il>

On 03/25/2010 06:42 PM, Michael Collins wrote:
> Ummm
> You're looking for the file "ronf-kicked.wav" - I'm pretty sure that's
> not what you want. Try "conf-kicked.wav" and see if that's any better....
> -MC
>
> On Wed, Mar 24, 2010 at 11:29 PM, eliimap <elihayun at gmail.com
> <mailto:elihayun at gmail.com>> wrote:
>
>     Hi
>
>     I set the variable "default_language" to be "ru" and then I run a
>     js to
>     play a file
>
>
>     ....
>
>     session.streamFile("conference/ronf-kicked.wav", "");
>
>     ....
>
>
>
>     I got an error message
>
>
>     2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File
>     [/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav]
>     [System error : No such file or directory.]
>
>
>     that indicates that mod_sndfile is not checking this value but always
>     using en
>
>
>     Please check
>
>
>     Eli
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
No, The file is OK. I did that on purpose to get the error so I will
know where the program is looking for the file.
When I am putting the correct file name I get it with ENGLISH and not
the language indicates by the variable.

Eli

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From cucku.cucku at yahoo.com.vn  Wed Mar 31 00:51:13 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Wed, 31 Mar 2010 15:51:13 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?q?V=E1=BB=81=3A__need_help_on_call_to_?=
 =?utf-8?q?gateway?=
In-Reply-To: <FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
Message-ID: <203894.53857.qm@web76213.mail.sg1.yahoo.com>

Hi David and Brian

sorry to bother you again
the domain name : sip.yeah.com is not real, How to fix the issue

FreeSwitch register to sip server with user 071234 successull
FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
the extension make call out using the sip server is fail
and the error :
sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
nua(0x9473108): call state changed: init -> calling, sent offer
soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
nua(0x9473108): event i_state INVITE sent
nua(0x9473108): event r_invite 503 DNS Error
nua(0x9473108): call state changed: calling -> init
nua(0x9473108): event i_state 503 DNS Error
nua(0x9473108): event i_terminated 503 DNS Error


i create 3 file:
    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call

the content of 071234.xml :
</include>
 <gateway name="071234">
  <param name="username" value="071234"/>
  <param name="from-user" value="071234"/>
  <param name="proxy" value="sip.yeah.com"/>
  <param name="username" value="071234"/>
  <param name="password" value="hadn"/>
  <param name="register-proxy" value="x.x.x.x"/>
  <param name="register" value="true"/>
  <param name="retry-seconds" value="10"/>
 </gateway>
</include>

the content of 00_inbound_did_fpt.xml:
<include>
  <extension name="public_did_fpt">
    <condition field="destination_number" expression="071234">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
</include>

the content of  01_fpt.net.xml:
<include>
  <extension name="sip.yeah.com">
    <condition field="destination_number" expression="^(\d+)$">
      <action application="bridge" data="sofia/gateway/071234/$1"/>
    </condition>
  </extension>
</include>



________________________________
T?: David Ponzone <david.ponzone at gmail.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010 
Ch? ??: Re: [Freeswitch-users] need help on call to gateway

False,

I think Brian is sleeping so I give you a quick answer.

For OUTBOUND:

Following my previous mail, you have to check if your user 1000 has the default context configured.
If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
(You may also remove from this path the remaining .xml files from the default install)
In 01_whatever.xml, put:


<include>
    <extension name="Route_to_sip.yeah.com">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="bridge" data="sofia/gateway/sip.yeah.com/0$1"/>
      </condition>
    </extension>

<include>

So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.

For INBOUND:

In conf/dialplan/public/, add a file named whatever.xml with the following content:

<include>
   <extension name="Inbound_from_sip.yeah.com">
      <condition field="destination_number" expression="^YOUR_DID$">
        <action application="transfer" data="1000 XML default"/>
      </condition>
    </extension>

</include>

Here, YOUR_DID is the DID number your provider allocated to you.
That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.

Of course, after doing all this, reloadxml.

It will perhaps not fit in your config exactly, but you should get the idea to get started.


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 30/03/2010 ? 09:01, false a ?crit :

Hi Brian
>
>could you guide me more to config on freeswitch
>
>i want xilte with user 1000 can make call out to sip server 
>and the prefix when make call out to sip server : 0
>and receive incomming call from SIP server
>
>what file name i should create 
>
>Thank you
>
>
>--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:
>
>
>>T?: Brian West <brian at freeswitch.org>
>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>??n: freeswitch-users at lists.freeswitch.org
>>Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>
>>
>>its looking in context default... not public.
>>
>>
>>/b
>>
>>
>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>
>>Hi all
>>>
>>>i use freeswitch to register to sip server
>>>the domain is not real: sip.yeah.com
>>>the outbound proxy : 118.69.145.5
>>>
>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>></include>
>>>    <gateway name="sip.yeah.com">
>>>        <param name="username" value="071234"/>
>>>        <param name="from-user" value="071234"/>    
>>>        <param name="proxy" value="sip.yeah.com"/>
>>>        <param name="username" value="071234"/>
>>>        <param name="password" value="1234"/>
>>>        <param name="register-proxy" value="118.69.145.5"/>
>>>        <param name="register" value="true"/>
>>>        <param name="retry-seconds" value="10"/>
>>>    </gateway>
>>></include>
>>>
>>>so i use xlite register to freeswitch with user 1000
>>>i start freeswitch , freeswitch start ok,
>>>freeswitch register to sip server successfull with username 071234
>>>xlite register to freeswitch successfull
>>>
>>>so i edit the public.xml
>>>for incomming call from sip server, i will forward to extension 1000
>>>    <extension name="sip.yeah.com">
>>>      <condition field="destination_number" expression="071234">
>>>        <action application="transfer" data="1000 XML default"/>
>>>      </condition>
>>>    </extension>
>>>
>>>but when i use xlite make call out from freeswitch to sip server and get no routing
>>> 
>> 
>________________________________
 T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo! Mail m?i h?m nay! _______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>



      L??t web nhanh h?n. Internet Explorer 8 t?i ?u h?a cho Yahoo!, t? ??ng kh?i ??ng 2 trang b?n th?ch m?i l?n m? tr?nh duy?t. T?i IE8 t?i ??y! http://downloads.yahoo.com/vn/internetexplorer/
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From david.ponzone at gmail.com  Wed Mar 31 01:07:54 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 31 Mar 2010 10:07:54 +0200
Subject: [Freeswitch-users] Newbie MWI question
In-Reply-To: <x2g87f2f3b91003301439ya29f1bc7y4207825ddf8f63d8@mail.gmail.com>
References: <8b61bd671003301241r5981479aucad295f15ac50cb3@mail.gmail.com>
	<v2s87f2f3b91003301345vb158fc2aj9592babe70e9b735@mail.gmail.com>
	<6358F826-7276-440A-A8D2-7D0AB6F75A6A@gmail.com>
	<x2g87f2f3b91003301439ya29f1bc7y4207825ddf8f63d8@mail.gmail.com>
Message-ID: <363010E3-4AC8-4757-B68D-423700BA8266@gmail.com>

Whaaaaattttt ?
FS does not talk MiNet ???
We need mod_minet :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/03/2010 ? 23:39, Michael Collins a ?crit :

>
>
> 2010/3/30 David Ponzone <david.ponzone at gmail.com>
> Michael,
>
> Yes, as many Mitel phones, I think they can do both MiNet protocol  
> and SIP.
> I dunno how good is their SIP though...
>
> Well, FreeSWITCH ne parles pas MiNet! :) You'll need to try SIP and  
> hope for the best.
> -MC
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 30/03/2010 ? 22:45, Michael Collins a ?crit :
>
>> Is that a SIP phone?
>>
>> On Tue, Mar 30, 2010 at 12:41 PM, lloyd thomas <lloydie.t at googlemail.com 
>> > wrote:
>> I am more or less using the default config files with FS. I have a  
>> Mitel 5212 phone, which I leave a message for but the MWI light  
>> never illuminates
>> Any Ideas?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Wed Mar 31 01:12:50 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 31 Mar 2010 10:12:50 +0200
Subject: [Freeswitch-users]
 =?utf-8?q?V=E1=BB=81=3A__need_help_on_call_to_?= =?utf-8?q?gateway?=
In-Reply-To: <203894.53857.qm@web76213.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
Message-ID: <25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>

This looks like a DNS error.
Are you sure you can resolve sip.yeah.com from the FS host ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 31/03/2010 ? 09:51, false a ?crit :

> Hi David and Brian
>
> sorry to bother you again
> the domain name : sip.yeah.com is not real, How to fix the issue
>
> FreeSwitch register to sip server with user 071234 successull
> FreeSwitch receive incomming from sip server and then transfer to  
> 1000 extension successfull
> the extension make call out using the sip server is fail
> and the error :
> sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
> nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
> nua(0x9473108): call state changed: init -> calling, sent offer
> soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c],  
> [(nil)]) called
> nua(0x9473108): event i_state INVITE sent
> nua(0x9473108): event r_invite 503 DNS Error
> nua(0x9473108): call state changed: calling -> init
> nua(0x9473108): event i_state 503 DNS Error
> nua(0x9473108): event i_terminated 503 DNS Error
>
> i create 3 file:
>     1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/ 
> external for register to sip server
>     1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/ 
> default for outbound call
>     1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/ 
> dialplan/public for incomming call
>
> the content of 071234.xml :
> </include>
>  <gateway name="071234">
>   <param name="username" value="071234"/>
>   <param name="from-user" value="071234"/>
>   <param name="proxy" value="sip.yeah.com"/>
>   <param name="username" value="071234"/>
>   <param name="password" value="hadn"/>
>   <param name="register-proxy" value="x.x.x.x"/>
>   <param name="register" value="true"/>
>   <param name="retry-seconds" value="10"/>
>  </gateway>
> </include>
>
> the content of 00_inbound_did_fpt.xml:
> <include>
>   <extension name="public_did_fpt">
>     <condition field="destination_number" expression="071234">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="transfer" data="1000 XML default"/>
>     </condition>
>   </extension>
> </include>
>
> the content of  01_fpt.net.xml:
> <include>
>   <extension name="sip.yeah.com">
>     <condition field="destination_number" expression="^(\d+)$">
>       <action application="bridge" data="sofia/gateway/071234/$1"/>
>     </condition>
>   </extension>
> </include>
>
> T?: David Ponzone <david.ponzone at gmail.com>
> ??n: freeswitch-users at lists.freeswitch.org
> G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010
> Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>
> False,
>
> I think Brian is sleeping so I give you a quick answer.
>
> For OUTBOUND:
>
> Following my previous mail, you have to check if your user 1000 has  
> the default context configured.
> If it does, then you just need to add a file named 01_whatever.xml  
> in conf/dialplan/default/.
> (You may also remove from this path the remaining .xml files from  
> the default install)
> In 01_whatever.xml, put:
>
> <include>
>     <extension name="Route_to_sip.yeah.com">
>       <condition field="destination_number" expression="^(\d+)$">
>         <action application="bridge" data="sofia/gateway/ 
> sip.yeah.com/0$1"/>
>       </condition>
>     </extension>
> <include>
>
> So when you dial any number not previously matched in default.xml  
> (local users, voicemail, ...), this will bridge it to your gateway,  
> adding 0 as a prefix.
>
> For INBOUND:
>
> In conf/dialplan/public/, add a file named whatever.xml with the  
> following content:
>
> <include>
>    <extension name="Inbound_from_sip.yeah.com">
>       <condition field="destination_number" expression="^YOUR_DID$">
>         <action application="transfer" data="1000 XML default"/>
>       </condition>
>     </extension>
> </include>
>
> Here, YOUR_DID is the DID number your provider allocated to you.
> That is an exact match, so if you're not sure about the exact  
> format, remove the leading ^, and match that on the longest part of  
> the number you know about.
>
> Of course, after doing all this, reloadxml.
>
> It will perhaps not fit in your config exactly, but you should get  
> the idea to get started.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 30/03/2010 ? 09:01, false a ?crit :
>
>> Hi Brian
>>
>> could you guide me more to config on freeswitch
>>
>> i want xilte with user 1000 can make call out to sip server
>> and the prefix when make call out to sip server : 0
>> and receive incomming call from SIP server
>>
>> what file name i should create
>>
>> Thank you
>>
>>
>> --- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ??  
>> vi?t:
>>
>> T?: Brian West <brian at freeswitch.org>
>> Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>> ??n: freeswitch-users at lists.freeswitch.org
>> Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>
>> its looking in context default... not public.
>>
>> /b
>>
>> On Mar 29, 2010, at 11:33 PM, false wrote:
>>
>>> Hi all
>>>
>>> i use freeswitch to register to sip server
>>> the domain is not real: sip.yeah.com
>>> the outbound proxy : 118.69.145.5
>>>
>>> i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/ 
>>> sip_profiles/external folder
>>> </include>
>>>     <gateway name="sip.yeah.com">
>>>         <param name="username" value="071234"/>
>>>         <param name="from-user" value="071234"/>
>>>         <param name="proxy" value="sip.yeah.com"/>
>>>         <param name="username" value="071234"/>
>>>         <param name="password" value="1234"/>
>>>         <param name="register-proxy" value="118.69.145.5"/>
>>>         <param name="register" value="true"/>
>>>         <param name="retry-seconds" value="10"/>
>>>     </gateway>
>>> </include>
>>>
>>> so i use xlite register to freeswitch with user 1000
>>> i start freeswitch , freeswitch start ok,
>>> freeswitch register to sip server successfull with username 071234
>>> xlite register to freeswitch successfull
>>>
>>> so i edit the public.xml
>>> for incomming call from sip server, i will forward to extension 1000
>>>     <extension name="sip.yeah.com">
>>>       <condition field="destination_number" expression="071234">
>>>         <action application="transfer" data="1000 XML default"/>
>>>       </condition>
>>>     </extension>
>>>
>>> but when i use xlite make call out from freeswitch to sip server  
>>> and get no routing
>>
>>
>> T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo!  
>> Mail m?i h?m nay!_______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
> T?o m?t n?i ?? chat tr?n blog l? chuy?n  
> nh?._______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From cucku.cucku at yahoo.com.vn  Wed Mar 31 01:22:29 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Wed, 31 Mar 2010 16:22:29 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?b?VuG7gTogIFbhu4E6ICBuZWVkIGhlbHAgb24g?=
 =?utf-8?q?call_to_gateway?=
In-Reply-To: <25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
Message-ID: <160047.29255.qm@web76212.mail.sg1.yahoo.com>

Hi David

the domain sip.yeah.com cannot resolve in the FS host
the thing here is FW support<param name="register-proxy" value="x.x.x.x"/> for Registration to remote SIP server
so the domain no need to resolve
how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,

Thank you




________________________________
T?: David Ponzone <david.ponzone at gmail.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 15:12:50, Th? T?, 31 th?ng 3 2010 
Ch? ??: Re: [Freeswitch-users] V?:  need help on call to gateway

This looks like a DNS error.
Are you sure you can resolve sip.yeah.com from the FS host ?


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/03/2010 ? 09:51, false a ?crit :

Hi David and Brian
>
>sorry to bother you again
>the domain name : sip.yeah.com is not real, How to fix the issue
>
>FreeSwitch register to sip server with user 071234 successull
>FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
>the extension make call out using the sip server is fail
>and the error :
>sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
>nua(0x9473108): call state changed: init -> calling, sent offer
>soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
>nua(0x9473108): event i_state INVITE sent
>nua(0x9473108): event r_invite 503 DNS Error
>nua(0x9473108): call state changed: calling -> init
>nua(0x9473108): event i_state 503 DNS Error
>nua(0x9473108): event i_terminated 503 DNS Error
>
>
>i create 3 file:
>    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
>    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
>    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call
>
>the content of 071234.xml :
></include>
> <gateway name="071234">
>  <param name="username" value="071234"/>
>  <param name="from-user" value="071234"/>
>  <param name="proxy" value="sip.yeah.com"/>
>  <param name="username" value="071234"/>
>  <param name="password" value="hadn"/>
>  <param name="register-proxy" value="x.x.x.x"/>
>  <param name="register" value="true"/>
>  <param name="retry-seconds" value="10"/>
> </gateway>
></include>
>
>the content of 00_inbound_did_fpt.xml:
><include>
>  <extension name="public_did_fpt">
>    <condition field="destination_number" expression="071234">
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="transfer" data="1000 XML default"/>
>    </condition>
>  </extension>
></include>
>
>the content of  01_fpt.net.xml:
><include>
>  <extension name="sip.yeah.com">
>    <condition field="destination_number" expression="^(\d+)$">
>      <action application="bridge" data="sofia/gateway/071234/$1"/>
>    </condition>
>  </extension>
></include>
>
>
>
________________________________
T?: David Ponzone <david.ponzone at gmail.com>
>??n: freeswitch-users at lists.freeswitch.org
>G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010 
>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>
>False,
>
>
>I think Brian is sleeping so I give you a quick answer.
>
>
>For OUTBOUND:
>
>
>Following my previous mail, you have to check if your user 1000 has the default context configured.
>If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
>(You may also remove from this path the remaining .xml files from the default install)
>In 01_whatever.xml, put:
>
>
><include>
>    <extension name="Route_to_sip.yeah.com">
>      <condition field="destination_number" expression="^(\d+)$">
>        <action application="bridge" data="sofia/gateway/sip.yeah.com/0$1"/>
>      </condition>
>    </extension>
>
><include>
>
>
>So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
>
>
>For INBOUND:
>
>
>In conf/dialplan/public/, add a file named whatever.xml with the following content:
>
>
><include>
>   <extension name="Inbound_from_sip.yeah.com">
>      <condition field="destination_number" expression="^YOUR_DID$">
>        <action application="transfer" data="1000 XML default"/>
>      </condition>
>    </extension>
>
></include>
>
>
>Here, YOUR_DID is the DID number your provider allocated to you.
>That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.
>
>
>Of course, after doing all this, reloadxml.
>
>
>It will perhaps not fit in your config exactly, but you should get the idea to get started.
>
>
>David Ponzone  Direction Technique
>email: david.ponzone at ipeva.fr
>tel:      01 74 03 18 97
>gsm:   06 66 98 76 34
>
>
>Service Client IPeva
>tel:      0811 46 26 26
>www.ipeva.fr  -   www.ipeva-studio.com
>
>
>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>Le 30/03/2010 ? 09:01, false a ?crit :
>
>Hi Brian
>>
>>could you guide me more to config on freeswitch
>>
>>i want xilte with user 1000 can make call out to sip server 
>>and the prefix when make call out to sip server : 0
>>and receive incomming call from SIP server
>>
>>what file name i should create 
>>
>>Thank you
>>
>>
>>--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:
>>
>>
>>>T?: Brian West <brian at freeswitch.org>
>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>??n: freeswitch-users at lists.freeswitch.org
>>>Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>>
>>>
>>>its looking in context default... not public.
>>>
>>>
>>>/b
>>>
>>>
>>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>>
>>>Hi all
>>>>
>>>>i use freeswitch to register to sip server
>>>>the domain is not real: sip.yeah.com
>>>>the outbound proxy : 118.69.145.5
>>>>
>>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>>></include>
>>>>    <gateway name="sip.yeah.com">
>>>>        <param name="username" value="071234"/>
>>>>        <param name="from-user" value="071234"/>    
>>>>        <param name="proxy" value="sip.yeah.com"/>
>>>>        <param name="username" value="071234"/>
>>>>        <param name="password" value="1234"/>
>>>>        <param name="register-proxy" value="118.69.145.5"/>
>>>>        <param name="register" value="true"/>
>>>>        <param name="retry-seconds" value="10"/>
>>>>    </gateway>
>>>></include>
>>>>
>>>>so i use xlite register to freeswitch with user 1000
>>>>i start freeswitch , freeswitch start ok,
>>>>freeswitch register to sip server successfull with username 071234
>>>>xlite register to freeswitch successfull
>>>>
>>>>so i edit the public.xml
>>>>for incomming call from sip server, i will forward to extension 1000
>>>>    <extension name="sip.yeah.com">
>>>>      <condition field="destination_number" expression="071234">
>>>>        <action application="transfer" data="1000 XML default"/>
>>>>      </condition>
>>>>    </extension>
>>>>
>>>>but when i use xlite make call out from freeswitch to sip server and get no routing
>>>> 
>>> 
>>________________________________
T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo! Mail m?i h?m nay!_______________________________________________
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>
>________________________________
Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
>T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?._______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>



      L??t web nhanh h?n. Internet Explorer 8 t?i ?u h?a cho Yahoo!, t? ??ng kh?i ??ng 2 trang b?n th?ch m?i l?n m? tr?nh duy?t. T?i IE8 t?i ??y! http://downloads.yahoo.com/vn/internetexplorer/
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From david.ponzone at gmail.com  Wed Mar 31 01:50:00 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 31 Mar 2010 10:50:00 +0200
Subject: [Freeswitch-users]
 =?utf-8?b?VuG7gTogIFbhu4E6ICBuZWVkIGhlbHAgb24g?=
 =?utf-8?q?call_to_gateway?=
In-Reply-To: <160047.29255.qm@web76212.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
Message-ID: <1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>

I guess you need to put the IP of the proxy also in:

   <param name="proxy" value="x.x.x.x"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 31/03/2010 ? 10:22, false a ?crit :

> Hi David
>
> the domain sip.yeah.com cannot resolve in the FS host
> the thing here is FW support <param name="register-proxy"  
> value="x.x.x.x"/> for Registration to remote SIP server
> so the domain no need to resolve
> how to do the same thing when call out, no need to resolve the  
> domain name: sip.yeah.com,
>
> Thank you
>
> T?: David Ponzone <david.ponzone at gmail.com>
> ??n: freeswitch-users at lists.freeswitch.org
> G?i ng?y: 15:12:50, Th? T?, 31 th?ng 3 2010
> Ch? ??: Re: [Freeswitch-users] V?: need help on call to gateway
>
> This looks like a DNS error.
> Are you sure you can resolve sip.yeah.com from the FS host ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 31/03/2010 ? 09:51, false a ?crit :
>
>> Hi David and Brian
>>
>> sorry to bother you again
>> the domain name : sip.yeah.com is not real, How to fix the issue
>>
>> FreeSwitch register to sip server with user 071234 successull
>> FreeSwitch receive incomming from sip server and then transfer to  
>> 1000 extension successfull
>> the extension make call out using the sip server is fail
>> and the error :
>> sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>> nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
>> nua(0x9473108): call state changed: init -> calling, sent offer
>> soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c],  
>> [(nil)]) called
>> nua(0x9473108): event i_state INVITE sent
>> nua(0x9473108): event r_invite 503 DNS Error
>> nua(0x9473108): call state changed: calling -> init
>> nua(0x9473108): event i_state 503 DNS Error
>> nua(0x9473108): event i_terminated 503 DNS Error
>>
>> i create 3 file:
>>     1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/ 
>> external for register to sip server
>>     1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/ 
>> default for outbound call
>>     1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/ 
>> dialplan/public for incomming call
>>
>> the content of 071234.xml :
>> </include>
>>  <gateway name="071234">
>>   <param name="username" value="071234"/>
>>   <param name="from-user" value="071234"/>
>>   <param name="proxy" value="sip.yeah.com"/>
>>   <param name="username" value="071234"/>
>>   <param name="password" value="hadn"/>
>>   <param name="register-proxy" value="x.x.x.x"/>
>>   <param name="register" value="true"/>
>>   <param name="retry-seconds" value="10"/>
>>  </gateway>
>> </include>
>>
>> the content of 00_inbound_did_fpt.xml:
>> <include>
>>   <extension name="public_did_fpt">
>>     <condition field="destination_number" expression="071234">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="transfer" data="1000 XML default"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> the content of  01_fpt.net.xml:
>> <include>
>>   <extension name="sip.yeah.com">
>>     <condition field="destination_number" expression="^(\d+)$">
>>       <action application="bridge" data="sofia/gateway/071234/$1"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> T?: David Ponzone <david.ponzone at gmail.com>
>> ??n: freeswitch-users at lists.freeswitch.org
>> G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010
>> Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>
>> False,
>>
>> I think Brian is sleeping so I give you a quick answer.
>>
>> For OUTBOUND:
>>
>> Following my previous mail, you have to check if your user 1000 has  
>> the default context configured.
>> If it does, then you just need to add a file named 01_whatever.xml  
>> in conf/dialplan/default/.
>> (You may also remove from this path the remaining .xml files from  
>> the default install)
>> In 01_whatever.xml, put:
>>
>> <include>
>>     <extension name="Route_to_sip.yeah.com">
>>       <condition field="destination_number" expression="^(\d+)$">
>>         <action application="bridge" data="sofia/gateway/ 
>> sip.yeah.com/0$1"/>
>>       </condition>
>>     </extension>
>> <include>
>>
>> So when you dial any number not previously matched in default.xml  
>> (local users, voicemail, ...), this will bridge it to your gateway,  
>> adding 0 as a prefix.
>>
>> For INBOUND:
>>
>> In conf/dialplan/public/, add a file named whatever.xml with the  
>> following content:
>>
>> <include>
>>    <extension name="Inbound_from_sip.yeah.com">
>>       <condition field="destination_number" expression="^YOUR_DID$">
>>         <action application="transfer" data="1000 XML default"/>
>>       </condition>
>>     </extension>
>> </include>
>>
>> Here, YOUR_DID is the DID number your provider allocated to you.
>> That is an exact match, so if you're not sure about the exact  
>> format, remove the leading ^, and match that on the longest part of  
>> the number you know about.
>>
>> Of course, after doing all this, reloadxml.
>>
>> It will perhaps not fit in your config exactly, but you should get  
>> the idea to get started.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ? l'intention exclusive de ses destinataires. Toute  
>> utilisation ou diffusion non autoris?e est interdite. Tout message  
>> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
>> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
>> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>>
>> Le 30/03/2010 ? 09:01, false a ?crit :
>>
>>> Hi Brian
>>>
>>> could you guide me more to config on freeswitch
>>>
>>> i want xilte with user 1000 can make call out to sip server
>>> and the prefix when make call out to sip server : 0
>>> and receive incomming call from SIP server
>>>
>>> what file name i should create
>>>
>>> Thank you
>>>
>>>
>>> --- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ??  
>>> vi?t:
>>>
>>> T?: Brian West <brian at freeswitch.org>
>>> Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>> ??n: freeswitch-users at lists.freeswitch.org
>>> Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>>
>>> its looking in context default... not public.
>>>
>>> /b
>>>
>>> On Mar 29, 2010, at 11:33 PM, false wrote:
>>>
>>>> Hi all
>>>>
>>>> i use freeswitch to register to sip server
>>>> the domain is not real: sip.yeah.com
>>>> the outbound proxy : 118.69.145.5
>>>>
>>>> i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/ 
>>>> sip_profiles/external folder
>>>> </include>
>>>>     <gateway name="sip.yeah.com">
>>>>         <param name="username" value="071234"/>
>>>>         <param name="from-user" value="071234"/>
>>>>         <param name="proxy" value="sip.yeah.com"/>
>>>>         <param name="username" value="071234"/>
>>>>         <param name="password" value="1234"/>
>>>>         <param name="register-proxy" value="118.69.145.5"/>
>>>>         <param name="register" value="true"/>
>>>>         <param name="retry-seconds" value="10"/>
>>>>     </gateway>
>>>> </include>
>>>>
>>>> so i use xlite register to freeswitch with user 1000
>>>> i start freeswitch , freeswitch start ok,
>>>> freeswitch register to sip server successfull with username 071234
>>>> xlite register to freeswitch successfull
>>>>
>>>> so i edit the public.xml
>>>> for incomming call from sip server, i will forward to extension  
>>>> 1000
>>>>     <extension name="sip.yeah.com">
>>>>       <condition field="destination_number" expression="071234">
>>>>         <action application="transfer" data="1000 XML default"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>> but when i use xlite make call out from freeswitch to sip server  
>>>> and get no routing
>>>
>>>
>>> T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo!  
>>> Mail m?i h?m nay!_______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
>> T?o m?t n?i ?? chat tr?n blog l? chuy?n  
>> nh?._______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> Yahoo! Mail nay NHANH H?N - Th? ngay! 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From hungngm at bkav.com.vn  Wed Mar 31 04:02:33 2010
From: hungngm at bkav.com.vn (=?utf-8?Q?Nguy=E1=BB=85n_M=E1=BA=A1nh_H=C3=B9ng__?=)
Date: Wed, 31 Mar 2010 18:02:33 +0700
Subject: [Freeswitch-users] Some question about mod_fifo ??
Message-ID: <9B583D983E27413B3A7D77EE29FE069F7D1F1FAC@hungngm>



 

Hi Mike and Seven Du.

Thanks to yours help.

I known the mechanism of mod_fifo.

>>
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
.

What a pity, It can't solve this problem. I can't use freeswitch for my call
center.

Hope new version can solve this !!!

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From tculjaga at gmail.com  Wed Mar 31 05:25:24 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 31 Mar 2010 14:25:24 +0200
Subject: [Freeswitch-users] Radius/Authentication/Authorization
In-Reply-To: <65d96fc81003301242w425a6aedk5d6d0ad2c1638edf@mail.gmail.com>
References: <COL109-W15576D4B6E4D7BFFFFEDD7F41F0@phx.gbl>
	<65d96fc81003301242w425a6aedk5d6d0ad2c1638edf@mail.gmail.com>
Message-ID: <g2w65d96fc81003310525g6994f140uf98d4a675b069a99@mail.gmail.com>

Abid, any progress ?

On Tue, Mar 30, 2010 at 9:42 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello,
>
> here is an example
>
>
>
> <configuration name="rad_auth.conf" description="radius authentification
> module">
>   <settings>
>      <!-- backward compatibility to allow radiusclient config file instead
> of an embedded config -->
>      <!--  <param name="radius_config"
> value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->
>   </settings>
>
>   <client>
>     <param name="authserver" value="10.1.1.10:1812:gateway"/>
>     <param name="dictionary"
> value="/usr/local/etc/radiusclient/dictionary.all"/>
>     <param name="seqfile" value="/var/run/radius.seq"/>
>     <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>
>     <param name="default_realm" value=""/>
>     <param name="radius_timeout" value="3"/>
>     <param name="radius_retries" value="2"/>
>     <param name="radius_deadtime" value="0"/>
>     <param name="bindaddr" value="*"/>
>   </client>
>
>   <vsas>
>     <!--
>          name:       just a description
>          value:      direct input or variable
>          pec:        vendor ID (0 for default, 9 for cisco...)
>          expr:       1 for channel variable, 2 for direct input
>          direction:  in for radius-request, out for radius-response
>     -->
>
>     <!-- mappings for radius request message; input attributes -->
>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
> direction="in"/>
>     <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9"
> expr="1" direction="in"/>
>     <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9"
> expr="1" direction="in"/>
>     <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0"
> expr="1" direction="in"/>
>     <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
> direction="in"/>
>     <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0"
> expr="0" direction="in"/>
>     <param name="Login-User" id="1" value="1" pec="0" expr="0"
> direction="in"/>
>
>     <!-- mappings for radius-response message; output values from returning
> outributes -->
>     <param name="BILING_MODEL" id="109" value="billing_model" pec="9"
> expr="0" direction="out"/>
>     <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9"
> expr="0" direction="out"/>
>     <param name="CURRENCY" id="110" value="currency" pec="9" expr="0"
> direction="out"/>
>     <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9"
> expr="0" direction="out"/>
>     <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0"
> direction="out"/>
>     <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration"
> pec="9" expr="0" direction="out"/>
>     <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9"
> expr="0" direction="out"/>
>     <!-- expr param is to be ignored here-->
>   </vsas>
> </configuration>
>
>
>
>
> in the dialplan you need to trigger auth as:
>
>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>
> there are two behaviours:
>
> 1. authorize the call according to username&pass and dialed number - if
> authorized, the radius server returns credit time towards the dialed number
> 2. authorize the call according to username&pass - if authorized, the
> radius server returns the current account balance
>
>
>
>
>
>
>   <extension name="unitest_rad-ANI-auth">
>     <condition field="destination_number" expression="^601$">
>       <action application="log" data="INFO  PRIJE RAD_AUTH "/>
>
>       <action inline="true" application="set"
> data="CALLID=h323-conf-id=${uuid}"/>
>       <action inline="true" application="set"
> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>       <action inline="true" application="set"
> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
> <!--      <action inline="true" application="set"
> data="CALLINGNUMBER=${caller_id_number}"/> -->
>       <action inline="true" application="set"
> data="CALLINGNUMBER=38516060333"/>
>       <action inline="true" application="set" data="USERNAME=38516060333"/>
> <!--      <action inline="true" application="set" data="USERNAME=209354"/>
> -->
>       <action inline="true" application="set" data="PASSWD=003282"/>
> <!--      <action inline="true" application="set"
> data="DIALED_NUMBER=16094191500"/>  -->
>
>       <action application="sleep" data="2000"/>
>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>
>       <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>       <action application="log" data="INFO
> billing_model=${billing_model}"/>
>       <action application="log" data="INFO
> credit_amount=${credit_amount}"/>
>       <action application="log" data="INFO  currency=${currency}"/>
>       <action application="log" data="INFO
> preffered_lang=${preffered_lang}"/>
>       <action application="log" data="INFO  credit_time=${credit_time}"/>
>       <action application="log" data="INFO
> h323_ivr_duration=${h323_ivr_duration}"/>
>       <action application="log" data="INFO  return_code=${return_code}"/>
>       <!-- <action application="execute_extension" data="AUTH XML
> default"/> -->
>     </condition>
>   </extension>
>
>
>   <extension name="unitest_rad-ANI-balance">
>     <condition field="destination_number" expression="^602$">
>       <action application="log" data="INFO  PRIJE RAD_AUTH "/>
>
>       <action inline="true" application="set"
> data="CALLID=h323-conf-id=${uuid}"/>
>       <action inline="true" application="set"
> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>       <action inline="true" application="set"
> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
> <!--      <action inline="true" application="set"
> data="CALLINGNUMBER=${caller_id_number}"/> -->
>       <action inline="true" application="set"
> data="CALLINGNUMBER=38516060333"/>
>       <action inline="true" application="set" data="USERNAME=38516060333"/>
> <!--      <action inline="true" application="set" data="USERNAME=209354"/>
> -->
>       <action inline="true" application="set" data="PASSWD=003282"/>
>       <action inline="true" application="set"
> data="DIALED_NUMBER=16094191500"/>
>
>       <action application="sleep" data="2000"/>
>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>
>       <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>       <action application="log" data="INFO
> billing_model=${billing_model}"/>
>       <action application="log" data="INFO
> credit_amount=${credit_amount}"/>
>       <action application="log" data="INFO  currency=${currency}"/>
>       <action application="log" data="INFO
> preffered_lang=${preffered_lang}"/>
>       <action application="log" data="INFO  credit_time=${credit_time}"/>
>       <action application="log" data="INFO
> h323_ivr_duration=${h323_ivr_duration}"/>
>       <action application="log" data="INFO  return_code=${return_code}"/>
>       <!-- <action application="execute_extension" data="AUTH XML
> default"/> -->
>     </condition>
>   </extension>
>
>
> will update the wiki by the end of the week.
>
>
> you have enough information for now.
>
> Tihomir.
>
>
>
> On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch at live.com>wrote:
>
>>  Hi Neal and other Contributors to FS,
>>
>> I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS
>> to download and install it by "make current" and it is successfully built
>> and installed. Could you please outline the detailed steps to do all this
>> configuration.
>>
>> If it is still not ready, is there any other method that somebody has
>> already implemented like perl scripts etc. Any help in this regard is badly
>> required. THanks for your cooperation.
>>
>> Regards
>> ----------
>> Abid Saleem
>> Sr. Product Manager
>>
>>
>> ------------------------------
>> Hotmail: Trusted email with powerful SPAM protection. Sign up now.<https://signup.live.com/signup.aspx?id=60969>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From cucku.cucku at yahoo.com.vn  Wed Mar 31 05:36:37 2010
From: cucku.cucku at yahoo.com.vn (false)
Date: Wed, 31 Mar 2010 20:36:37 +0800 (SGT)
Subject: [Freeswitch-users] =?utf-8?b?VuG7gTogIFbhu4E6ICBW4buBOiAgbmVlZCBo?=
 =?utf-8?q?elp_on_call_to_gateway?=
In-Reply-To: <1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
Message-ID: <938433.73668.qm@web76209.mail.sg1.yahoo.com>

Hi David

when change the <param name="proxy" value="x.x.x.x"/>
the Register message send out with wrong format : from sip:071234 at x.x.x.x
the right format : sip:071234 at sip.yeah.com

is there any way to fix it

Thank you





________________________________
T?: David Ponzone <david.ponzone at gmail.com>
??n: freeswitch-users at lists.freeswitch.org
G?i ng?y: 15:50:00, Th? T?, 31 th?ng 3 2010 
Ch? ??: Re: [Freeswitch-users] V?:  V?:  need help on call to gateway

I guess you need to put the IP of the proxy also in:


  <param name="proxy" value="x.x.x.x"/>


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 31/03/2010 ? 10:22, false a ?crit :

Hi David
>
>the domain sip.yeah.com cannot resolve in the FS host
>the thing here is FW support <param name="register-proxy" value="x.x.x.x"/> for Registration to remote SIP server
>so the domain no need to resolve
>how to do the same thing when call out, no need to resolve the domain name: sip.yeah.com,
>
>Thank you
>
>
>
>
________________________________
T?: David Ponzone <david.ponzone at gmail.com>
>??n: freeswitch-users at lists.freeswitch.org
>G?i ng?y: 15:12:50, Th? T?, 31 th?ng 3 2010 
>Ch? ??: Re: [Freeswitch-users] V?: need help on call to gateway
>
>This looks like a DNS error.
>Are you sure you can resolve sip.yeah.com from the FS host ?
>
>
>David Ponzone  Direction Technique
>email: david.ponzone at ipeva.fr
>tel:      01 74 03 18 97
>gsm:   06 66 98 76 34
>
>
>Service Client IPeva
>tel:      0811 46 26 26
>www.ipeva.fr  -   www.ipeva-studio.com
>
>
>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
>
>Le 31/03/2010 ? 09:51, false a ?crit :
>
>Hi David and Brian
>>
>>sorry to bother you again
>>the domain name : sip.yeah.com is not real, How to fix the issue
>>
>>FreeSwitch register to sip server with user 071234 successull
>>FreeSwitch receive incomming from sip server and then transfer to 1000 extension successfull
>>the extension make call out using the sip server is fail
>>and the error :
>>sres_cache_get(0x93b4538, A, "sip.yeah.com.") returned 1 entries
>>nta: for "sip.yeah.com" query "sip.yeah.com" A (cached)
>>nua(0x9473108): call state changed: init -> calling, sent offer
>>soa_get_local_sdp(static::0xb7892a68, [0xb774b080], [0xb774b07c], [(nil)]) called
>>nua(0x9473108): event i_state INVITE sent
>>nua(0x9473108): event r_invite 503 DNS Error
>>nua(0x9473108): call state changed: calling -> init
>>nua(0x9473108): event i_state 503 DNS Error
>>nua(0x9473108): event i_terminated 503 DNS Error
>>
>>
>>i create 3 file:
>>    1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for register to sip server
>>    1 file 01_fpt.net.xml in usr/local/freeswitch/conf/dialplan/default for outbound call
>>    1 file 00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for incomming call
>>
>>the content of 071234.xml :
>></include>
>> <gateway name="071234">
>>  <param name="username" value="071234"/>
>>  <param name="from-user" value="071234"/>
>>  <param name="proxy" value="sip.yeah.com"/>
>>  <param name="username" value="071234"/>
>>  <param name="password" value="hadn"/>
>>  <param name="register-proxy" value="x.x.x.x"/>
>>  <param name="register" value="true"/>
>>  <param name="retry-seconds" value="10"/>
>> </gateway>
>></include>
>>
>>the content of 00_inbound_did_fpt.xml:
>><include>
>>  <extension name="public_did_fpt">
>>    <condition field="destination_number" expression="071234">
>>      <action application="set" data="domain_name=$${domain}"/>
>>      <action application="transfer" data="1000 XML default"/>
>>    </condition>
>>  </extension>
>></include>
>>
>>the content of  01_fpt.net.xml:
>><include>
>>  <extension name="sip.yeah.com">
>>    <condition field="destination_number" expression="^(\d+)$">
>>      <action application="bridge" data="sofia/gateway/071234/$1"/>
>>    </condition>
>>  </extension>
>></include>
>>
>>
>>
________________________________
T?: David Ponzone <david.ponzone at gmail.com>
>>??n: freeswitch-users at lists.freeswitch.org
>>G?i ng?y: 14:26:13, Th? Ba, 30 th?ng 3 2010 
>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>
>>False,
>>
>>
>>I think Brian is sleeping so I give you a quick answer.
>>
>>
>>For OUTBOUND:
>>
>>
>>Following my previous mail, you have to check if your user 1000 has the default context configured.
>>If it does, then you just need to add a file named 01_whatever.xml in conf/dialplan/default/.
>>(You may also remove from this path the remaining .xml files from the default install)
>>In 01_whatever.xml, put:
>>
>>
>><include>
>>    <extension name="Route_to_sip.yeah.com">
>>      <condition field="destination_number" expression="^(\d+)$">
>>        <action application="bridge" data="sofia/gateway/sip.yeah.com/0$1"/>
>>      </condition>
>>    </extension>
>>
>><include>
>>
>>
>>So when you dial any number not previously matched in default.xml (local users, voicemail, ...), this will bridge it to your gateway, adding 0 as a prefix.
>>
>>
>>For INBOUND:
>>
>>
>>In conf/dialplan/public/, add a file named whatever.xml with the following content:
>>
>>
>><include>
>>   <extension name="Inbound_from_sip.yeah.com">
>>      <condition field="destination_number" expression="^YOUR_DID$">
>>        <action application="transfer" data="1000 XML default"/>
>>      </condition>
>>    </extension>
>>
>></include>
>>
>>
>>Here, YOUR_DID is the DID number your provider allocated to you.
>>That is an exact match, so if you're not sure about the exact format, remove the leading ^, and match that on the longest part of the number you know about.
>>
>>
>>Of course, after doing all this, reloadxml.
>>
>>
>>It will perhaps not fit in your config exactly, but you should get the idea to get started.
>>
>>
>>David Ponzone  Direction Technique
>>email: david.ponzone at ipeva.fr
>>tel:      01 74 03 18 97
>>gsm:   06 66 98 76 34
>>
>>
>>Service Client IPeva
>>tel:      0811 46 26 26
>>www.ipeva.fr  -   www.ipeva-studio.com
>>
>>
>>Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>>
>>
>>Le 30/03/2010 ? 09:01, false a ?crit :
>>
>>Hi Brian
>>>
>>>could you guide me more to config on freeswitch
>>>
>>>i want xilte with user 1000 can make call out to sip server 
>>>and the prefix when make call out to sip server : 0
>>>and receive incomming call from SIP server
>>>
>>>what file name i should create 
>>>
>>>Thank you
>>>
>>>
>>>--- Ng?y Th? 3, 30/03/10, Brian West <brian at freeswitch.org> ?? vi?t:
>>>
>>>
>>>>T?: Brian West <brian at freeswitch.org>
>>>>Ch? ??: Re: [Freeswitch-users] need help on call to gateway
>>>>??n: freeswitch-users at lists.freeswitch.org
>>>>Ng?y: Th? Ba, 30 th?ng 3, 2010, 4:37
>>>>
>>>>
>>>>its looking in context default... not public.
>>>>
>>>>
>>>>/b
>>>>
>>>>
>>>>On Mar 29, 2010, at 11:33 PM, false wrote:
>>>>
>>>>Hi all
>>>>>
>>>>>i use freeswitch to register to sip server
>>>>>the domain is not real: sip.yeah.com
>>>>>the outbound proxy : 118.69.145.5
>>>>>
>>>>>i create the sip.yeah.com.xml in /usr/local/freeswitch/conf/sip_profiles/external folder
>>>>></include>
>>>>>    <gateway name="sip.yeah.com">
>>>>>        <param name="username" value="071234"/>
>>>>>        <param name="from-user" value="071234"/>    
>>>>>        <param name="proxy" value="sip.yeah.com"/>
>>>>>        <param name="username" value="071234"/>
>>>>>        <param name="password" value="1234"/>
>>>>>        <param name="register-proxy" value="118.69.145.5"/>
>>>>>        <param name="register" value="true"/>
>>>>>        <param name="retry-seconds" value="10"/>
>>>>>    </gateway>
>>>>></include>
>>>>>
>>>>>so i use xlite register to freeswitch with user 1000
>>>>>i start freeswitch , freeswitch start ok,
>>>>>freeswitch register to sip server successfull with username 071234
>>>>>xlite register to freeswitch successfull
>>>>>
>>>>>so i edit the public.xml
>>>>>for incomming call from sip server, i will forward to extension 1000
>>>>>    <extension name="sip.yeah.com">
>>>>>      <condition field="destination_number" expression="071234">
>>>>>        <action application="transfer" data="1000 XML default"/>
>>>>>      </condition>
>>>>>    </extension>
>>>>>
>>>>>but when i use xlite make call out from freeswitch to sip server and get no routing
>>>>> 
>>>> 
>>>________________________________
T?t h?n, tho?ng g?n h?n, nhanh h?n - Tr?i nghi?m Yahoo! Mail m?i h?m nay!_______________________________________________
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>
>>________________________________
Thi?t k? ngay m?t Pingbox ??c ??o cho ri?ng b?n!
>>T?o m?t n?i ?? chat tr?n blog l? chuy?n nh?._______________________________________________
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>
>________________________________
Yahoo! Mail nay NHANH H?N - Th? ngay!_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>



      C?m gi?c an to?n h?n khi online. N?ng c?p l?n Internet Explorer 8 m?i, an to?n h?n,t?i ?u h?a cho Yahoo!. Ho?n to?n mi?n ph?. T?i IE8 t?i ??y! 
http://downloads.yahoo.com/vn/internetexplorer/
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From dftoro at yahoo.com  Wed Mar 31 05:53:41 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 31 Mar 2010 05:53:41 -0700 (PDT)
Subject: [Freeswitch-users] Variable "default_language" has no effect on
	javascript
In-Reply-To: <4BB2D616.1080109@savion.huji.ac.il>
Message-ID: <362373.80871.qm@web33501.mail.mud.yahoo.com>

Hi, set sound_prefix variable ( http://wiki.freeswitch.org/wiki/Channel_Variables#sound_prefix )

Diego Toro

http://lacarretade.blogspot.com/

--- On Tue, 3/30/10, Eli Hayun <elihayun at gmail.com> wrote:

From: Eli Hayun <elihayun at gmail.com>
Subject: Re: [Freeswitch-users] Variable "default_language" has no effect on javascript
To: "freeswitch-users at lists.freeswitch.org" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, March 30, 2010, 11:56 PM





  
On 03/25/2010 06:42 PM, Michael Collins wrote:
Ummm

You're looking for the file "ronf-kicked.wav" - I'm pretty sure that's
not what you want. Try "conf-kicked.wav" and see if that's any
better....

-MC

  

  On Wed, Mar 24, 2010 at 11:29 PM, eliimap <elihayun at gmail.com>
wrote:

  Hi

    

I set the variable "default_language" to be "ru" and then I run a js to

play a file

    

    

....

    

session.streamFile("conference/ronf-kicked.wav", "");

    

....

    

    

    

I got an error message

    

    

2010-03-25 10:23:46.430391 [ERR] mod_sndfile.c:194 Error Opening File

[/freeswitch-1.0.5-20100314-0400/sounds/en/us/callie/conference/ronf-kicked.wav]

[System error : No such file or directory.]

    

    

that indicates that mod_sndfile is not checking this value but always

using en

    

    

Please check

    

    

Eli

    

    

_______________________________________________

FreeSWITCH-users mailing list

    FreeSWITCH-users at lists.freeswitch.org

    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

    http://www.freeswitch.org

  
  
  


No, The file is OK. I did that on purpose to get the error so I will
know where the program is looking for the file.

When I am putting the correct file name I get it with ENGLISH and not
the language indicates by the variable.



Eli



 

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dome at tel.co.th  Wed Mar 31 06:17:58 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 31 Mar 2010 20:17:58 +0700
Subject: [Freeswitch-users] Regular Expression for match ip address
In-Reply-To: <B0DB56DE1A764F10AB32AB2CDD1A2D99@MOBILEE1705>
References: <8ccbff061003291205i47746d23h2f70812c83e88c48@mail.gmail.com>
	<9B82C0908BDD4F7D88AD14C4403631FA@MOBILEE1705>
	<8ccbff061003300949k7d6d4b7eo48577062cc9a4e8f@mail.gmail.com>
	<B0DB56DE1A764F10AB32AB2CDD1A2D99@MOBILEE1705>
Message-ID: <m2r8ccbff061003310617lb635e772u15ab05ad1b8da35d@mail.gmail.com>

It's work perfect :)   I love Freeswitch...

Dome C.

2010/3/31 Madovsky <infos at madovsky.org>:
> if you want this
> <node type="deny" cidr="0.0.0.0/0"/>
>
> so use <list name="test" default="deny">
> and use allow inside
>
> ----- Original Message -----
> From: "Dome Charoenyost" <dome at tel.co.th>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, March 30, 2010 12:49 PM
> Subject: Re: [Freeswitch-users] Regular Expression for match ip address
>
>
>> Thanks All.
>> How to deny all and allow some ?
>> I try
>> ? ?<list name="test" default="allow">
>> ? ? ? ?<node type="deny" cidr="0.0.0.0/0"/>
>> ? ? ? ?<node type="allow" cidr="10.0.0.1/32"/>
>> ? ? ? ?<node type="allow" cidr="115.178.1.0/24"/>
>> ? ?</list>
>>
>> but don't work.
>>
>> Dome C.
>>
>> 2010/3/30 Madovsky <infos at madovsky.org>:
>>> you need to use ACL in autoload_configs/acl.conf.xml
>>>
>>> ----- Original Message -----
>>> From: "Dome Charoenyost" <dome at tel.co.th>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, March 29, 2010 3:05 PM
>>> Subject: [Freeswitch-users] Regular Expression for match ip address
>>>
>>>
>>>> Dear All,
>>>> I want to autorize incoming call to public context by IP by
>>>> <condition field="network_addr" expression="^192\.168\.1\.1$"/>
>>>> but some time i want to allow 1 Class C or some range of IP. How can do
>>>> that ?
>>>>
>>>> BG
>>>>
>>>> Dome C.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Wed Mar 31 06:31:12 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Wed, 31 Mar 2010 20:31:12 +0700
Subject: [Freeswitch-users] I need some comment for Priority Call
Message-ID: <o2j8ccbff061003310631g5a885c74ra67d2a1ac3eb3ee@mail.gmail.com>

Dear All,
    I'm planing to make priority call in dial plan. it's mean if we
have maximum 30 concurrent port for dial out.
and use all port. when some extention make call (VIP call) dial plan
hangup some longest call and give port for VIP.


any idea ?

Dome C.


From devel at thom.fr.eu.org  Wed Mar 31 06:51:17 2010
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 31 Mar 2010 15:51:17 +0200
Subject: [Freeswitch-users]
 =?utf-8?b?VuG7gTogIFbhu4E6ICBW4buBOiAgbmVlZCBo?=
 =?utf-8?q?elp_on_call_to_gateway?=
In-Reply-To: <938433.73668.qm@web76209.mail.sg1.yahoo.com>
References: <319878.5002.qm@web76204.mail.sg1.yahoo.com>
	<FA50ED89-A509-42C8-A889-64E8D0CE5A19@gmail.com>
	<203894.53857.qm@web76213.mail.sg1.yahoo.com>
	<25CA4E70-9596-4011-A1E0-42439E266E76@gmail.com>
	<160047.29255.qm@web76212.mail.sg1.yahoo.com>
	<1CD70388-E70D-46F5-8D9C-4C6585E14688@gmail.com>
	<938433.73668.qm@web76209.mail.sg1.yahoo.com>
Message-ID: <c0d94ea412f51451bc32f5ac4b409150@thom.fr.eu.org>



Then either provide DNS to that system so that it can resolve the
address, either create a static binding in /etc/hosts file on that host for
sip.yeah.com 

On Wed, 31 Mar 2010 20:36:37 +0800 (SGT), false  wrote:   Hi
David

when change the 
the Register message send out with wrong format :
from sip:071234 at x.x.x.x
the right format : sip:071234 at sip.yeah.com

is
there any way to fix it

Thank you

-------------------------
 T?: David
Ponzone 
??N: freeswitch-users at lists.freeswitch.org
G?I NGY: 15:50:00, Th?
T?, 31 th?ng 3 2010 
CH? ??: Re: [Freeswitch-users] V?: V?: need help on
call to gateway

I guess you need to put the IP of the proxy also in:      
                  David Ponzone Direction Technique email:
david.ponzone at ipeva.fr [1] tel: 01 74 03 18 97 gsm: 06 66 98 76 34 

Service Client IPeva  tel: 0811 46 26 26  www.ipeva.fr -
www.ipeva-studio.com 
  _Ce message et toutes les pi?ces jointes sont
confidentiels et ?tablis ? l'intention exclusive de ses destinataires.
Toute utilisation ou diffusion non autoris?e est interdite. Tout message
?lectronique est susceptible d'alt?ration. __IPEVA__ d?cline toute
responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou
falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le
d?truire imm?diatement et d'avertir l'exp?diteur._ 

  Le 31/03/2010 ?
10:22, false a ?crit : 
   Hi David

the domain sip.yeah.com cannot resolve
in the FS host
the thing here is FW support  for Registration to remote SIP
server
so the domain no need to resolve
how to do the same thing when call
out, no need to resolve the domain name: sip.yeah.com,

Thank
you

-------------------------
 T?: David Ponzone 
??N:
freeswitch-users at lists.freeswitch.org [3]
G?I NGY: 15:12:50, Th? T?, 31
th?ng 3 2010 
CH? ??: Re: [Freeswitch-users] V?: need help on call to
gateway

This looks like a DNS error. Are you sure you can resolve
sip.yeah.com from the FS host ? 
    David Ponzone Direction Technique
email: david.ponzone at ipeva.fr [4] tel: 01 74 03 18 97 gsm: 06 66 98 76 34 

Service Client IPeva  tel: 0811 46 26 26  www.ipeva.fr -
www.ipeva-studio.com 
  _Ce message et toutes les pi?ces jointes sont
confidentiels et ?tablis ? l'intention exclusive de ses destinataires.
Toute utilisation ou diffusion non autoris?e est interdite. Tout message
?lectronique est susceptible d'alt?ration. __IPEVA__ d?cline toute
responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou
falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le
d?truire imm?diatement et d'avertir l'exp?diteur._ 

  Le 31/03/2010 ?
09:51, false a ?crit : 
    Hi David and Brian

sorry to bother you
again
the domain name : sip.yeah.com is not real, How to fix the
issue

FreeSwitch register to sip server with user 071234
successull
FreeSwitch receive incomming from sip server and then transfer
to 1000 extension successfull
the extension make call out using the sip
server is fail
and the error :
sres_cache_get(0x93b4538, A,
"sip.yeah.com.") returned 1 entries
nta: for "sip.yeah.com" query
"sip.yeah.com" A (cached)
nua(0x9473108): call state changed: init ->
calling, sent offer
soa_get_local_sdp(static::0xb7892a68, [0xb774b080],
[0xb774b07c], [(nil)]) called
nua(0x9473108): event i_state INVITE
sent
nua(0x9473108): event r_invite 503 DNS Error
nua(0x9473108): call
state changed: calling -> init
nua(0x9473108): event i_state 503 DNS
Error
nua(0x9473108): event i_terminated 503 DNS Error

 i create 3 file:

1 file 071234.xml in /usr/local/freeswitch/conf/sip_profiles/external for
register to sip server
 1 file 01_fpt.net.xml in
usr/local/freeswitch/conf/dialplan/default for outbound call
 1 file
00_inbound_did_fpt.xml in /usr/local/freeswitch/conf/dialplan/public for
incomming call

the content of 071234.xml :

the content of
00_inbound_did_fpt.xml:

the content of
01_fpt.net.xml:

-------------------------
 T?: David Ponzone 
??N:
freeswitch-users at lists.freeswitch.org [6]
G?I NGY: 14:26:13, Th? Ba, 30
th?ng 3 2010 
CH? ??: Re: [Freeswitch-users] need help on call to
gateway

False,  I think Brian is sleeping so I give you a quick answer. 
For OUTBOUND:  Following my previous mail, you have to check if your user
1000 has the default context configured. If it does, then you just need to
add a file named 01_whatever.xml in conf/dialplan/default/. (You may also
remove from this path the remaining .xml files from the default install) In
01_whatever.xml, put:           

          So when you dial any number not
previously matched in default.xml (local users, voicemail, ...), this will
bridge it to your gateway, adding 0 as a prefix.  For INBOUND:  In
conf/dialplan/public/, add a file named whatever.xml with the following
content:           

         Here, YOUR_DID is the DID number your
provider allocated to you. That is an exact match, so if you're not sure
about the exact format, remove the leading ^, and match that on the longest
part of the number you know about.  Of course, after doing all this,
reloadxml.  It will perhaps not fit in your config exactly, but you should
get the idea to get started. 
    David Ponzone Direction Technique email:
david.ponzone at ipeva.fr [7] tel: 01 74 03 18 97 gsm: 06 66 98 76 34 

Service Client IPeva  tel: 0811 46 26 26  www.ipeva.fr -
www.ipeva-studio.com 
  _Ce message et toutes les pi?ces jointes sont
confidentiels et ?tablis ? l'intention exclusive de ses destinataires.
Toute utilisation ou diffusion non autoris?e est interdite. Tout message
?lectronique est susceptible d'alt?ration. __IPEVA__ d?cline toute
responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou
falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le
d?truire imm?diatement et d'avertir l'exp?diteur._ 

  Le 30/03/2010 ?
09:01, false a ?crit : 

 Hi Brian

could you guide me more to config on
freeswitch

i want xilte with user 1000 can make call out to sip server

and the prefix when make call out to sip server : 0
and receive incomming
call from SIP server

what file name i should create 

Thank you

--- Ng?y
TH? 3, 30/03/10, BRIAN WEST __ ?? vi?t:

T?: Brian West 
Ch? ??: Re:
[Freeswitch-users] need help on call to gateway
??n:
freeswitch-users at lists.freeswitch.org [9]
Ng?y: Th? Ba, 30 th?ng 3, 2010,
4:37

 its looking in context default... not public.  /b 
  On Mar 29,
2010, at 11:33 PM, false wrote: 

 Hi all

i use freeswitch to register to
sip server
the domain is not real: sip.yeah.com [10]
the outbound proxy :
118.69.145.5

i create the sip.yeah.com.xml in
/usr/local/freeswitch/conf/sip_profiles/external folder

 sip.yeah.com">


sip.yeah.com"/>

so i use xlite register to freeswitch with user 1000
i
start freeswitch , freeswitch start ok,
freeswitch register to sip server
successfull with username 071234
xlite register to freeswitch
successfull

so i edit the public.xml
for incomming call from sip server, i
will forward to extension 1000
 sip.yeah.com">

but when i use xlite make
call out from freeswitch to sip server and get no
routing

-------------------------
 T?t h?n, tho?ng g?n h?n, nhanh h?n -
Tr?i nghi?m Yahoo! Mail m?i h?m nay!
[11]_______________________________________________
FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[12]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[13]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[14]
http://www.freeswitch.org [15]

-------------------------
 Thi?t k?
ngay m?t Pingbox ??c ??o cho ri?ng b?n! [16]
T?o m?t n?i ?? chat tr?n blog
l? chuy?n
nh?._______________________________________________
FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[17]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[18]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[19]
http://www.freeswitch.org [20]       

-------------------------

Yahoo! Mail nay NHANH H?N - Th? ngay!
[21]_______________________________________________
FreeSWITCH-users
mailing list
FreeSWITCH-users at lists.freeswitch.org
[22]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[23]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[24]
http://www.freeswitch.org [25]       

-------------------------

Yahoo! Mail nay nhanh v? nhi?u kh?ng gian tho?ng h?n. H?y tr?i nghi?m ngay
h?m nay!  [26] 

 

Links:
------
[1] mailto:david.ponzone at ipeva.fr
[2]
mailto:david.ponzone at gmail.com
[3]
mailto:freeswitch-users at lists.freeswitch.org
[4]
mailto:david.ponzone at ipeva.fr
[5] mailto:david.ponzone at gmail.com
[6]
mailto:freeswitch-users at lists.freeswitch.org
[7]
mailto:david.ponzone at ipeva.fr
[8] mailto:brian at freeswitch.org
[9]
mailto:freeswitch-users at lists.freeswitch.org
[10] http://sip.yeah.com
[11]
http://vn.mail.yahoo.com
[12]
mailto:FreeSWITCH-users at lists.freeswitch.org
[13]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[14]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[15]
http://www.freeswitch.org
[16]
http://sg.rd.yahoo.com/vn/messenger/pingbox/mailtagline/*http://vn.messenger.yahoo.com/pingbox/
[17]
mailto:FreeSWITCH-users at lists.freeswitch.org
[18]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[19]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[20]
http://www.freeswitch.org
[21] http://vn.mail.yahoo.com
[22]
mailto:FreeSWITCH-users at lists.freeswitch.org
[23]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[24]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[25]
http://www.freeswitch.org
[26] http://vn.mail.yahoo.com
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From dujinfang at gmail.com  Wed Mar 31 07:10:14 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 31 Mar 2010 22:10:14 +0800
Subject: [Freeswitch-users] Some question about mod_fifo ??
In-Reply-To: <9B583D983E27413B3A7D77EE29FE069F7D1F1FAC@hungngm>
References: <9B583D983E27413B3A7D77EE29FE069F7D1F1FAC@hungngm>
Message-ID: <t2o23f91031003310710na0345f36yc1b647176f899a82@mail.gmail.com>

As discussed in the list, it's not a freeswitch problem but a reality of life.

Think about customer A and B calls in one after another, then if
FreeSWITCH call agent X with caller id A and Y with caller id B, and
angent Y answers before X, then

1) if bridge Y with A with the FIFO rule, then the caller id is wrong
2) if bridge Y with B, the caller id is right but it breaks the rule
of FIFO - A should be served before B!!  And what even worse is that
if X never answer A then A never can be served which is really
unfair!!

Of course you don't want 1), and you don't need mod_fifo if you want
behavior  2), you just need some dialplan trick or some simple Lua
script I think. Also FreeSWITCH is designed to be easily extended with
almost any languages so feel free to implement anything.

2010/3/31 Nguy?n M?nh H?ng <hungngm at bkav.com.vn>:
> Hi Mike and Seven Du.
> Thanks?to yours help.
> I known the mechanism of mod_fifo.
>>>http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html.
> What a pity, It?can't solve this problem. I can't use freeswitch for my call
> center.
> Hope new version can solve this !!!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tomb at cachecomm.com  Wed Mar 31 08:04:04 2010
From: tomb at cachecomm.com (Tom)
Date: Wed, 31 Mar 2010 09:04:04 -0600
Subject: [Freeswitch-users] {Scanned} Re: {Scanned} Calling Phone Behind
 Netgear Router (with NAT)
In-Reply-To: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>
References: <9BDC3BB8F7F147E9AA7D0843DBCD55EB@greyhawk.tonecommander.com>
Message-ID: <4BB36464.3030406@cachecomm.com>

Jerry Richards wrote:
> We are trying call a phone that resides behind a Netgear router (with NAT).
> The Bria softphone registers and we can make calls from the softphone, but
> we cannot make calls to the softphone.
>
>                 WAN
> ------------  public IP -------- 10.10.10.xx ----------------
> |Freeswitch|<---------->|Router|<----------->|Bria softphone|
> ------------            --------             ----------------
>
> Does anyone know of good documentation/example showing this configuration?
>
> Best Regards,
> Jerry
>
>
>
>   
Jerry,
Had the same problem with the 338 router and a 114. Even after the beta 
firmware it is was broken. I replaced them with linux firewalls problem gone


Tom



From justin at ejtown.org  Wed Mar 31 08:14:25 2010
From: justin at ejtown.org (Justin B Newman)
Date: Wed, 31 Mar 2010 11:14:25 -0400
Subject: [Freeswitch-users] bridge from-user
Message-ID: <s2y27b0d5bc1003310814t41f6bf95g48d755026204b2f8@mail.gmail.com>

I have tried a number of variations on:

<action application="bridge"
data="[from-user=abc,sip_auth_username=abc,sip_auth_password=123]sofia/external/$1 at voip.mydomain.com"/>

So far, to no avail.  In all cases, the From does not contain "abc",
but rather, the Caller-ID data.

FreeSWITCH Version 1.0.5-20100329-0400 (17128M)

Is it possible to modify From when placing an outbound call in absence
of a gateway?

-jbn


From neilp at cs.stanford.edu  Wed Mar 31 08:22:46 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Wed, 31 Mar 2010 08:22:46 -0700
Subject: [Freeswitch-users] debugging no audio I/O
In-Reply-To: <e5503bdb1003301134i794d0331w161518f05dcab7d@mail.gmail.com>
References: <e5503bdb1003292021s4cfd58e7n839ee74dc52330fa@mail.gmail.com> 
	<52013EC0-78D9-47BE-ADE0-CECF1B074F06@freeswitch.org>
	<e5503bdb1003292119n648e5930i5877a3ae71480def@mail.gmail.com> 
	<e5503bdb1003300914s3ed627cdsc31bb116466a1c4c@mail.gmail.com> 
	<6b65470d1003300926u3e85555ej942c84d021a7dbb7@mail.gmail.com> 
	<6b65470d1003300927v7646585bh7c4e231dd7a86dcd@mail.gmail.com> 
	<e5503bdb1003301134i794d0331w161518f05dcab7d@mail.gmail.com>
Message-ID: <x2ye5503bdb1003310822me12b771xb92abff4f3e4d222@mail.gmail.com>

Could this be a firewall issue? I have opened up ports 5060 and 5080 on the
server manually using iptables, which I don't believe I had to do on the old
server. What other ports should be opened of the ones listed
here<http://wiki.freeswitch.org/wiki/Firewall>,
especially for audio I/O? For reference netstat output is pasted below.

Thanks,
Neil

==================
[root at topaz ~]# netstat -tupln
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address               Foreign Address
State       PID/Program name
tcp        0      0 127.0.0.1:2208              0.0.0.0:*
LISTEN      3217/hpiod
tcp        0      0 128.32.226.175:5060         0.0.0.0:*
LISTEN      16609/freeswitch
tcp        0      0 0.0.0.0:3306                0.0.0.0:*
LISTEN      19228/mysqld
tcp        0      0 0.0.0.0:111                 0.0.0.0:*
LISTEN      2906/portmap
tcp        0      0 127.0.0.1:8021              0.0.0.0:*
LISTEN      16609/freeswitch
tcp        0      0 127.0.0.1:631               0.0.0.0:*
LISTEN      18870/cupsd
tcp        0      0 128.32.226.175:5080         0.0.0.0:*
LISTEN      16609/freeswitch
tcp        0      0 127.0.0.1:25                0.0.0.0:*
LISTEN      3288/sendmail: acce
tcp        0      0 0.0.0.0:1021                0.0.0.0:*
LISTEN      2959/rpc.statd
tcp        0      0 127.0.0.1:2207              0.0.0.0:*
LISTEN      3222/python
tcp        0      0 ::1:5060                    :::*
 LISTEN      16609/freeswitch
tcp        0      0 :::80                       :::*
 LISTEN      22999/httpd
tcp        0      0 :::22                       :::*
 LISTEN      19266/sshd
tcp        0      0 :::443                      :::*
 LISTEN      22999/httpd
udp        0      0 128.32.226.175:5060         0.0.0.0:*
            16609/freeswitch
udp        0      0 0.0.0.0:68                  0.0.0.0:*
            2809/dhclient
udp        0      0 128.32.226.175:5080         0.0.0.0:*
            16609/freeswitch
udp        0      0 0.0.0.0:5353                0.0.0.0:*
            3389/avahi-daemon:
udp        0      0 0.0.0.0:47338               0.0.0.0:*
            3389/avahi-daemon:
udp        0      0 0.0.0.0:111                 0.0.0.0:*
            2906/portmap
udp        0      0 0.0.0.0:631                 0.0.0.0:*
            18870/cupsd
udp        0      0 0.0.0.0:1015                0.0.0.0:*
            2959/rpc.statd
udp        0      0 0.0.0.0:1018                0.0.0.0:*
            2959/rpc.statd
udp        0      0 ::1:5060                    :::*
           16609/freeswitch
udp        0      0 :::5353                     :::*
           3389/avahi-daemon:
udp        0      0 :::43892                    :::*
           3389/avahi-daemon:

On Tue, Mar 30, 2010 at 11:34 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Old (Ubuntu) working FS install:
> FreeSWITCH Version 1.0.trunk (15859)
>
> New (CentOS) broken FS install:
> FreeSWITCH Version 1.0.trunk (17071)
>
> On Tue, Mar 30, 2010 at 9:27 AM, William Suffill <
> william.suffill at gmail.com> wrote:
>
>> Running fs_cli then running version also works.
>> --
>> freeswitch at internal> version
>>
>> FreeSWITCH Version 1.0.4 (exported)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Wed Mar 31 08:33:07 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Mar 2010 10:33:07 -0500
Subject: [Freeswitch-users] bridge from-user
In-Reply-To: <s2y27b0d5bc1003310814t41f6bf95g48d755026204b2f8@mail.gmail.com>
References: <s2y27b0d5bc1003310814t41f6bf95g48d755026204b2f8@mail.gmail.com>
Message-ID: <334AF8A2-5C7C-44E1-B8DB-82E47D8C301E@freeswitch.org>

Please read do_invite in sofia_glue.c  (someone needs to document these)


Have fun with these:


sip_invite_domain
sip_transport
sip_contact_user
sip_invite_call_id
sip_from_uri
sip_from_display
sip_invite_from_uri
sip_invite_to_uri
sip_invite_to_params
sip_invite_params
sip_invite_contact_params
sip_invite_from_params
sip_invite_req_uri


/b

On Mar 31, 2010, at 10:14 AM, Justin B Newman wrote:

> I have tried a number of variations on:
> 
> <action application="bridge"
> data="[from-user=abc,sip_auth_username=abc,sip_auth_password=123]sofia/external/$1 at voip.mydomain.com"/>
> 
> So far, to no avail.  In all cases, the From does not contain "abc",
> but rather, the Caller-ID data.
> 
> FreeSWITCH Version 1.0.5-20100329-0400 (17128M)
> 
> Is it possible to modify From when placing an outbound call in absence
> of a gateway?
> 
> -jbn



From adam.meehan at gmail.com  Tue Mar 30 18:57:27 2010
From: adam.meehan at gmail.com (Adam Meehan)
Date: Wed, 31 Mar 2010 12:57:27 +1100
Subject: [Freeswitch-users] mod_unimrcp with Nuance "Timed out waiting for
	channel to be ready"
Message-ID: <4BB2AC07.8040301@gmail.com>

I'm using mod_unimrcp module with the Nuance Speech Server 5.1 and I'm getting 
the "Timed out waiting for channel to be ready" when trying to use TTS. This is 
using MRCP v2 and FS trunk rev 17126.

I've not used MRCP and I am stuck. Any suggestions?

Thanks
Adam


From infos at madovsky.org  Wed Mar 31 08:27:38 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 31 Mar 2010 11:27:38 -0400
Subject: [Freeswitch-users] bridge from-user
References: <s2y27b0d5bc1003310814t41f6bf95g48d755026204b2f8@mail.gmail.com>
Message-ID: <4F0E53C2EA2744BEA890E59F47E09648@MOBILEE1705>

I think this line is wrong
sofia/external/



----- Original Message ----- 
From: "Justin B Newman" <justin at ejtown.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, March 31, 2010 11:14 AM
Subject: [Freeswitch-users] bridge from-user


>I have tried a number of variations on:
>
> <action application="bridge"
> data="[from-user=abc,sip_auth_username=abc,sip_auth_password=123]sofia/external/$1 at voip.mydomain.com"/>
>
> So far, to no avail.  In all cases, the From does not contain "abc",
> but rather, the Caller-ID data.
>
> FreeSWITCH Version 1.0.5-20100329-0400 (17128M)
>
> Is it possible to modify From when placing an outbound call in absence
> of a gateway?
>
> -jbn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From justin at ejtown.org  Wed Mar 31 08:49:53 2010
From: justin at ejtown.org (Justin B Newman)
Date: Wed, 31 Mar 2010 11:49:53 -0400
Subject: [Freeswitch-users] bridge from-user
In-Reply-To: <334AF8A2-5C7C-44E1-B8DB-82E47D8C301E@freeswitch.org>
References: <s2y27b0d5bc1003310814t41f6bf95g48d755026204b2f8@mail.gmail.com>
	<334AF8A2-5C7C-44E1-B8DB-82E47D8C301E@freeswitch.org>
Message-ID: <k2n27b0d5bc1003310849i9f3bd23bk73396ed18a798b7f@mail.gmail.com>

On Wed, Mar 31, 2010 at 11:33 AM, Brian West <brian at freeswitch.org> wrote:
> Please read do_invite in sofia_glue.c ?(someone needs to document these)
>
> sip_invite_from_uri

Many thanks for the pointer, both to the variable and to the correct
section of code.

In order to assert a specific From value in an INVITE, sans-gateway,
the correct statement, for me, is:

 <action application="bridge"
data="[sip_invite_from_uri=sip:abc,sip_auth_username=abc,sip_auth_password=def]sofia/external/$1 at voip.mydomain.com"/>

-jbn


From Suneel.Papineni at mettoni.com  Wed Mar 31 09:15:24 2010
From: Suneel.Papineni at mettoni.com (Suneel Papineni)
Date: Wed, 31 Mar 2010 17:15:24 +0100
Subject: [Freeswitch-users] Adding Record-Route Header
Message-ID: <3181A30B8C35AB4AA8577B78DDF4613806ADD2F7@nickel.mettonigroup.com>

Hi,

 

How can I add Record-Route (pointing to another FS acting as Proxy
server) for the calls handled by to FSComm client. 

 

Also can I set this while making a call or accepting a call using event
socket commands (pa answer / pa call xxxx). 

 

Thanks & Regards

Suneel

 

 


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*************************************************************************
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are addressed. If you have received this email in error please notify
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From brian at freeswitch.org  Wed Mar 31 09:22:20 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Mar 2010 11:22:20 -0500
Subject: [Freeswitch-users] Adding Record-Route Header
In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613806ADD2F7@nickel.mettonigroup.com>
References: <3181A30B8C35AB4AA8577B78DDF4613806ADD2F7@nickel.mettonigroup.com>
Message-ID: <AB29A4E9-7620-4FE7-883A-607CA17BD161@freeswitch.org>

You really have no need to do this because FreeSWITCH is not a proxy.  If it were a true proxy then a record route would make sense.

/b

On Mar 31, 2010, at 11:15 AM, Suneel Papineni wrote:

> Hi,
>  
> How can I add Record-Route (pointing to another FS acting as Proxy server) for the calls handled by to FSComm client.
>  
> Also can I set this while making a call or accepting a call using event socket commands (pa answer / pa call xxxx).
>  
> Thanks & Regards
> Suneel

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From jmesquita at freeswitch.org  Wed Mar 31 10:01:21 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 31 Mar 2010 14:01:21 -0300
Subject: [Freeswitch-users] Adding Record-Route Header
In-Reply-To: <AB29A4E9-7620-4FE7-883A-607CA17BD161@freeswitch.org>
References: <3181A30B8C35AB4AA8577B78DDF4613806ADD2F7@nickel.mettonigroup.com>
	<AB29A4E9-7620-4FE7-883A-607CA17BD161@freeswitch.org>
Message-ID: <ebbc2d551003311001v14c61df1qef785e5b8b1534a2@mail.gmail.com>

Maybe if you try to explain what you are trying to accomplish and why it
would make it easier for us to help.

Regards,
Jo?o Mesquita


On Wed, Mar 31, 2010 at 1:22 PM, Brian West <brian at freeswitch.org> wrote:

> You really have no need to do this because FreeSWITCH is not a proxy.  If
> it were a true proxy then a record route would make sense.
>
> /b
>
> On Mar 31, 2010, at 11:15 AM, Suneel Papineni wrote:
>
> Hi,
>
> How can I add Record-Route (pointing to another FS acting as Proxy server)
> for the calls handled by to FSComm client.
>
> Also can I set this while making a call or accepting a call using event
> socket commands (pa answer / pa call xxxx).
>
> Thanks & Regards
> Suneel
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kenfulmer at icstechnologysolutions.com  Wed Mar 31 10:33:08 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Wed, 31 Mar 2010 12:33:08 -0500
Subject: [Freeswitch-users] SIP Call Trace
Message-ID: <00be01cad0f8$3aff09e0$b0fd1da0$@com>

Perhaps I've missed it, but can someone explain how to capture SIP and SDP
messages for troubleshooting? 

 

Thanks,

 

Ken Fulmer

 

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From william.suffill at gmail.com  Wed Mar 31 10:44:18 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 31 Mar 2010 13:44:18 -0400
Subject: [Freeswitch-users] SIP Call Trace
In-Reply-To: <00be01cad0f8$3aff09e0$b0fd1da0$@com>
References: <00be01cad0f8$3aff09e0$b0fd1da0$@com>
Message-ID: <m2q6b65470d1003311044r88b7025lc049675876286c30@mail.gmail.com>

sofia allows you to enable siptrace from the cli
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP

Or external tools like tcpdump/ngrep etc can be used
http://wiki.freeswitch.org/wiki/Packet_Capture
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From troy at tlainvestments.com  Wed Mar 31 10:50:20 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Wed, 31 Mar 2010 10:50:20 -0700
Subject: [Freeswitch-users] SIP Call Trace
In-Reply-To: <00be01cad0f8$3aff09e0$b0fd1da0$@com>
References: <00be01cad0f8$3aff09e0$b0fd1da0$@com>
Message-ID: <E3F3232C-3EC2-4361-ACA0-72F9EABA283D@tlainvestments.com>

I use ngrep-sip (on linux) at http://dev.sipdoc.net/projects/sip-stuff/wiki/Ngrep-SIP

It sits on top of ngrep.  To get it to work on my installations, I have had to uncomment the interface=eth0 line and comment the #interface=any line in edit /etc/ngrep-sip.conf, but after that, it works swimmingly.

-Troy

On Mar 31, 2010, at 10:33 AM, Ken Fulmer wrote:

> Perhaps I?ve missed it, but can someone explain how to capture SIP and SDP messages for troubleshooting?
>  
> Thanks,
>  
> Ken Fulmer
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From cmrienzo at gmail.com  Wed Mar 31 10:59:24 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 31 Mar 2010 13:59:24 -0400
Subject: [Freeswitch-users] mod_unimrcp with Nuance "Timed out waiting
	for channel to be ready"
In-Reply-To: <4BB2AC07.8040301@gmail.com>
References: <4BB2AC07.8040301@gmail.com>
Message-ID: <i2n3687b0701003311059x797014c6y1c6b023f12b6a3f3@mail.gmail.com>

I don't remember which error conditions cause that problem.  Can you show me
a trace of your call and the MRCP profile you used?


On Tue, Mar 30, 2010 at 9:57 PM, Adam Meehan <adam.meehan at gmail.com> wrote:

> I'm using mod_unimrcp module with the Nuance Speech Server 5.1 and I'm
> getting
> the "Timed out waiting for channel to be ready" when trying to use TTS.
> This is
> using MRCP v2 and FS trunk rev 17126.
>
> I've not used MRCP and I am stuck. Any suggestions?
>
> Thanks
> Adam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lloyd.aloysius at gmail.com  Wed Mar 31 12:36:25 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 31 Mar 2010 15:36:25 -0400
Subject: [Freeswitch-users] svn up then make current - Errors
Message-ID: <v2i8a19bf2e1003311236l47e97deem3c38e0578c5f515b@mail.gmail.com>

Hello,

Trying to update the svn source.

svn up

make current

I got the following Errors.


cd libs/openzap && autoconf
configure.ac:9: error: possibly undefined macro: AM_INIT_AUTOMAKE
      If this token and others are legitimate, please use m4_pattern_allow.
      See the Autoconf documentation.
configure.ac:14: error: possibly undefined macro: AM_PROG_CC_C_O
configure.ac:41: error: possibly undefined macro: AC_PROG_LIBTOOL
configure.ac:119: error: possibly undefined macro: AM_CONDITIONAL
make: *** [libs/openzap/Makefile] Error 1


How to fix the make error.

Thanks
Lloyd
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From carlos.talbot at gmail.com  Wed Mar 31 12:45:15 2010
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Wed, 31 Mar 2010 14:45:15 -0500
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <20100313085113.GA32174@jdc.jasonjgw.net>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
	<20100313085113.GA32174@jdc.jasonjgw.net>
Message-ID: <p2t5800526b1003311245i9fa904fcl568576ca4840c75c@mail.gmail.com>

This no longer works. For some reason the Makefile was deleted in svn 17057.
I suppose this has to do with the GIT transition? I tried to build it with
the top level configure but got this:

=== configuring in libs/silk (/mnt/disk2/freeswitch_dev/libs/silk)
configure: running /bin/bash ./configure.gnu --disable-option-checking
'--prefix=/mnt/disk2/freeswitch'  'CONFIGURE_CFLAGS=-g -O2'
'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' --cache-file=/dev/null
--srcdir=.
./configure.gnu: line 3: ./configure: No such file or directory
configure: error: ./configure.gnu failed for libs/silk


On Sat, Mar 13, 2010 at 2:51 AM, Jason White <jason at jasonjgw.net> wrote:

> Code has now been posted at https://developer.skype.com/silk/
>
> It appears from that page that there may be patent issues for anything
> other
> than testing of the codec, although it is stated elsewhere that the licence
> terms are royalty-free.
>
> To build the module (for those planning to do the aforementioned testing)
> you
> need to run make in libs/silk first to compile the library, then build
> FreeSWITCH; this will change once the build system is updated
> appropriately.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Mar 31 13:01:27 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 31 Mar 2010 15:01:27 -0500
Subject: [Freeswitch-users] SILK Codec
In-Reply-To: <p2t5800526b1003311245i9fa904fcl568576ca4840c75c@mail.gmail.com>
References: <82160CC4-25E5-453F-8FD6-4889F9902682@freeswitch.org>
	<ec2ce17e1003081727u450b307m64b1512001584a5f@mail.gmail.com>
	<23f91031003091949n7c598fa9vf714a6b477ab539f@mail.gmail.com>
	<20100310051344.GA19653@jdc.jasonjgw.net>
	<23f91031003092137p689fb8c9nfb05c5613ef5e663@mail.gmail.com>
	<20100313085113.GA32174@jdc.jasonjgw.net>
	<p2t5800526b1003311245i9fa904fcl568576ca4840c75c@mail.gmail.com>
Message-ID: <E892EC87-DA57-48E4-9067-04E5FBAA47F8@freeswitch.org>

Its been auto-tooled and you'll have to rebootstrap.

/b

On Mar 31, 2010, at 2:45 PM, Carlos Talbot wrote:

> This no longer works. For some reason the Makefile was deleted in svn 17057. I suppose this has to do with the GIT transition? I tried to build it with the top level configure but got this:
> 
> === configuring in libs/silk (/mnt/disk2/freeswitch_dev/libs/silk)
> configure: running /bin/bash ./configure.gnu --disable-option-checking '--prefix=/mnt/disk2/freeswitch'  'CONFIGURE_CFLAGS=-g -O2' 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' --cache-file=/dev/null --srcdir=.
> ./configure.gnu: line 3: ./configure: No such file or directory
> configure: error: ./configure.gnu failed for libs/silk
> 



From jayesh.voip at gmail.com  Wed Mar 31 13:05:29 2010
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Thu, 1 Apr 2010 01:35:29 +0530
Subject: [Freeswitch-users] domain-wise context
Message-ID: <y2m4fe8b8271003311305xfa657666kc217e0b03069b702@mail.gmail.com>

Hi All,
I am quite very new to freeswitch and I am kind of playing with it to
understand it better.
I am primarily using Opensips as registrar and SIP Proxy and intend to use
FS as media server handling voicemails, IVR, Announcements, MeetMe etc. My
Opensips is a multi-domain setup and I wish to have all the configuration of
media-capabilties segregated domain-wise in the FS too.

For eg: When a call for user at domain1.com needs to go to voicemail, I
redirect that call to FS IP address keeping the URI intact. I add the
mailbox number as a header as X-Mailbox and have FS extract it and go to
appropriate mailbox. Similarly when a call for user at domain2.com needs to go
to voicemail I do the same thing.
The requirement is I want to maintain the dialplans for each domains
separately. Thus if call from Opensips comes to FS with domain as domain1,
the call should go to dialplan context domain1 and similarly if call from
Opensips comes to FS with domain2 the call handling should be mentioned in
the domain2 context.

The problem is; I am not able to send the calls to respective contexts
according to their domains when they come from Opensips. I've read the
examples on multi-domain setup and have tried taking some help from that
example, but whenever the call comes from Opensips to FS, it tries to go
into the context that is defined in the SIP Profile. If i don't mention
anything in the SIP Profile, it tries to search for default context.
I have tried the following:
1) Created file domain1.xml and domain2.xml in the directory folder.
2) mentioned parameters in domain1.xml as follows:
<include>
<domain name = "domain1.com"
<params>
<param name="user_context" value "domain1.com"/>
</params>
</domain>
</include>
3) Similarly done for file domain2.xml.

But I am just not able to get the calls to the required context according to
the domain value in the r-uri. In asterisk something like this in sip.conf
worked fine for me:
domain=domain1.com, domain1.com
domain=domain2.com, domain2.com
Can someone please help me understanding where I am going wrong or have I
mis-understood something?

Thanks in advance !!

--- Jayesh
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From stevendt at primrosebank.net  Wed Mar 31 13:42:16 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Wed, 31 Mar 2010 21:42:16 +0100
Subject: [Freeswitch-users] Phone Registration Events
Message-ID: <F4ABFD51F4A446F890D64E8568BFA2D3@bp1.ad.bp.com>

 
Hi,

Is there a log option to turn on logging of SIP registration messages ?

I have a mixture of phones and am having a problem with some of them periodically losing registration with FreeSWITCH.

All the phones start off working after provisioning and have registration expiry times configured of 3600 seconds.

I have tried changing the expiry time to no effect and I wanted to see if the phones were actually trying to renew the registration when they should.

Regards
Dave
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From david.ponzone at gmail.com  Wed Mar 31 13:57:53 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 31 Mar 2010 22:57:53 +0200
Subject: [Freeswitch-users] Phone Registration Events
In-Reply-To: <F4ABFD51F4A446F890D64E8568BFA2D3@bp1.ad.bp.com>
References: <F4ABFD51F4A446F890D64E8568BFA2D3@bp1.ad.bp.com>
Message-ID: <B4E72760-2732-48E5-A735-9BD48273D72D@gmail.com>

I would recommend taking SIP captures with wireshark for this.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 31/03/2010 ? 22:42, Dave Stevenson a ?crit :

>
> Hi,
>
> Is there a log option to turn on logging of SIP registration  
> messages ?
>
> I have a mixture of phones and am having a problem with some of them  
> periodically losing registration with FreeSWITCH.
>
> All the phones start off working after provisioning and have  
> registration expiry times configured of 3600 seconds.
>
> I have tried changing the expiry time to no effect and I wanted to  
> see if the phones were actually trying to renew the registration  
> when they should.
>
> Regards
> Dave
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adam.meehan at gmail.com  Wed Mar 31 15:25:13 2010
From: adam.meehan at gmail.com (Adam Meehan)
Date: Thu, 01 Apr 2010 09:25:13 +1100
Subject: [Freeswitch-users] mod_unimrcp with Nuance "Timed out waiting
 for channel to be ready"
In-Reply-To: <i2n3687b0701003311059x797014c6y1c6b023f12b6a3f3@mail.gmail.com>
References: <4BB2AC07.8040301@gmail.com>
	<i2n3687b0701003311059x797014c6y1c6b023f12b6a3f3@mail.gmail.com>
Message-ID: <4BB3CBC9.9000506@gmail.com>

On 1/04/10 4:59 AM, Christopher Rienzo wrote:
> I don't remember which error conditions cause that problem.  Can you
> show me a trace of your call and the MRCP profile you used?

Here is the trace

http://pastebin.freeswitch.org/12595

I have just used the default Nuance V2 profile that comes with FS.

http://pastebin.freeswitch.org/12596

I've turned off iptables in case of any firewall issues on both the FS and 
Nuance server.

Thanks for the help.
Adam



From marketing at cluecon.com  Wed Mar 31 15:47:12 2010
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 31 Mar 2010 15:47:12 -0700
Subject: [Freeswitch-users] ClueCon MMX - It Will Be Here Before You Know It!
Message-ID: <x2y87f2f3b91003311547ze816e9afq73b06e55c7e0409f@mail.gmail.com>

Greetings!

We wanted to remind everyone to get a head start on their ClueCon
arrangements. We have great things planned for this year and we don't want
anyone to miss out. Please call 1-877-742-CLUE (2583) to register or if you
have any questions at all about ClueCon.

See you in August!

-ClueCon Team
http://www.cluecon.com
877.742.CLUE
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From msc at freeswitch.org  Wed Mar 31 16:03:57 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Mar 2010 16:03:57 -0700
Subject: [Freeswitch-users] Call Recording & bind_meta_app
In-Reply-To: <7bcfdd291003291726g4b12a6bau7b990b2fae4d17b2@mail.gmail.com>
References: <7bcfdd291003291726g4b12a6bau7b990b2fae4d17b2@mail.gmail.com>
Message-ID: <j2k87f2f3b91003311603sf4b253afrd956b437d42455eb@mail.gmail.com>

On Mon, Mar 29, 2010 at 5:26 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I need to add the ability to record a call to an existing
> infrastructure, and I'm intending to use FreeSwitch to do this. The
> recording needs to happen upon a DTMF command, which I understand is
> normally done via bind_meta_app. From what I can tell, bind_meta_app
> requires that you use the * key in addition to some other key.
> Unfortunately, the system is already doing something else with *, so
> that's not a key I can use to get this to work.
>
> My question, then: what is the best way to record a call using DTMFs
> that do not include *?
>
Use tone_detect and pick the frequencies for the key you prefer to use. More
info:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

DTMF frequencies are listed here:
http://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling#Keypad


> - Should I have Freeswitch forward an appropriate DTMF to a
> controlling application that starts recording?
>
> - Should I try to modify the bind_meta_app code to use another digit
> instead (say, #)?  If so, where is the right place to look for this?


> As a separate, but related question, I also need to play a disclaimer
> when the recording starts ("This call is now being recorded."). Is it
> OK to use bind_meta_app to bind 2 apps to the same DTMF combo?
>

This part is easy: just have the destination app be "execute_extension" like
in the Local_Extension bind_meta_app examples.
-MC
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From cmrienzo at gmail.com  Wed Mar 31 18:09:39 2010
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 31 Mar 2010 21:09:39 -0400
Subject: [Freeswitch-users] mod_unimrcp with Nuance "Timed out waiting
	for channel to be ready"
In-Reply-To: <4BB3CBC9.9000506@gmail.com>
References: <4BB2AC07.8040301@gmail.com>
	<i2n3687b0701003311059x797014c6y1c6b023f12b6a3f3@mail.gmail.com>
	<4BB3CBC9.9000506@gmail.com>
Message-ID: <u2w3687b0701003311809uc0e1eeb8qc7b978cc832da12@mail.gmail.com>

Looks like unimrcp sent the SIP INVITE to the MRCP server and never received
a reply.  The INVITE's SDP has a loopback address in there, so I think you
need to explicitly set the "client-ip" and "rtp-ip" instead of using auto.



On Wed, Mar 31, 2010 at 6:25 PM, Adam Meehan <adam.meehan at gmail.com> wrote:

> On 1/04/10 4:59 AM, Christopher Rienzo wrote:
> > I don't remember which error conditions cause that problem.  Can you
> > show me a trace of your call and the MRCP profile you used?
>
> Here is the trace
>
> http://pastebin.freeswitch.org/12595
>
> I have just used the default Nuance V2 profile that comes with FS.
>
> http://pastebin.freeswitch.org/12596
>
> I've turned off iptables in case of any firewall issues on both the FS and
> Nuance server.
>
> Thanks for the help.
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adam.meehan at gmail.com  Wed Mar 31 19:14:42 2010
From: adam.meehan at gmail.com (Adam Meehan)
Date: Thu, 01 Apr 2010 13:14:42 +1100
Subject: [Freeswitch-users] mod_unimrcp with Nuance "Timed out waiting
 for channel to be ready"
In-Reply-To: <u2w3687b0701003311809uc0e1eeb8qc7b978cc832da12@mail.gmail.com>
References: <4BB2AC07.8040301@gmail.com>	<i2n3687b0701003311059x797014c6y1c6b023f12b6a3f3@mail.gmail.com>	<4BB3CBC9.9000506@gmail.com>
	<u2w3687b0701003311809uc0e1eeb8qc7b978cc832da12@mail.gmail.com>
Message-ID: <4BB40192.20905@gmail.com>


> Looks like unimrcp sent the SIP INVITE to the MRCP server and never
> received a reply.  The INVITE's SDP has a loopback address in there, so
> I think you need to explicitly set the "client-ip" and "rtp-ip" instead
> of using auto.

Yes that did the trick. I also had to explicit set the IP on the Nuance server. 
I wasn't paying attention to which server would be using the IP being 
transmitted in the trace. Makes perfect sense now.

I'm now getting an error from the MRCP client on hangup, but I will see I can 
solve it myself. Otherwise I will start another thread.

Thank you
Adam



From serget68 at msn.com  Wed Mar 31 23:03:38 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Thu, 1 Apr 2010 01:03:38 -0500
Subject: [Freeswitch-users] ZRTP on the FreeSWITCH PBX - questions on the
	log report
Message-ID: <COL108-W1015593859EC5279008F8CD91D0@phx.gbl>


I recently deployed ZRTP on the FreeSWITCH PBX on a Linux machine and I have two X-Lite softphones (on Windows) properly connected and secure with zfone and I am wondering why the FreeSWITCH log is returning the following errors. Also, the zfone on the machine receiving calls won't stay secure for a longtime after the two softphones are connected. I need help to understand this.
 
Thank you!
 
 
2010-03-28 21:40:11.870118 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.890334 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.910739 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.930733 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.950166 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.970874 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:11.990705 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.128549 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.130517 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.150576 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.170183 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.190531 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9
2010-03-28 21:40:12.210543 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:13.799105 [NOTICE] sofia.c:4789 Hangup sofia/internal/1000 at 192.168.118.178 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2010-03-28 21:40:13.811844 [NOTICE] switch_ivr_originate.c:3079 Hangup sofia/internal/sip:1001 at 192.168.118.128:12430 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2010-03-28 21:40:13.811844 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]
2010-03-28 21:40:13.811844 [INFO] mod_dptools.c:2353 Originate Failed. Cause: ORIGINATOR_CANCEL
2010-03-28 21:40:13.863496 [NOTICE] switch_core_session.c:1179 Session 17 (sofia/internal/1000 at 192.168.118.178) Ended
2010-03-28 21:40:13.863496 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1000 at 192.168.118.178 [CS_DESTROY]
2010-03-28 21:40:13.935508 [NOTICE] switch_core_session.c:1179 Session 18 (sofia/internal/sip:1001 at 192.168.118.128:12430) Ended
2010-03-28 21:40:13.935508 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/sip:1001 at 192.168.118.128:12430 [CS_DESTROY]
2010-03-28 21:40:35.542292 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001 at 192.168.118.178 [d34e988a-82b4-4891-8c19-82a695eb8cc2]
2010-03-28 21:40:35.562382 [INFO] mod_dialplan_xml.c:418 Processing 1001->1000 in context default
2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 1 execute_extension::dx XML features
2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1001.2010-03-28-21-40-35.wav
2010-03-28 21:40:35.562382 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 3 execute_extension::cf XML features
2010-03-28 21:40:35.562382 [NOTICE] switch_channel.c:669 New Channel sofia/internal/sip:1000 at 192.168.118.179:22558 [bd9f7fa3-426a-4cbe-8a67-9c531d14bba0]
2010-03-28 21:40:36.064353 [INFO] sofia.c:662 Update Callee ID to "1000" <1000>
2010-03-28 21:40:36.085286 [NOTICE] sofia.c:4223 Ring-Ready sofia/internal/sip:1000 at 192.168.118.179:22558!
2010-03-28 21:40:36.138043 [INFO] switch_ivr_originate.c:1064 Sending early media
2010-03-28 21:40:36.139955 [NOTICE] mod_sofia.c:1907 Pre-Answer sofia/internal/1001 at 192.168.118.178!
2010-03-28 21:40:36.310490 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.423250 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.430517 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.451432 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.574454 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.590511 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:36.610380 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:56.354321 [NOTICE] sofia.c:4733 Channel [sofia/internal/sip:1000 at 192.168.118.179:22558] has been answered
2010-03-28 21:40:56.410969 [NOTICE] switch_ivr_originate.c:3103 Channel [sofia/internal/1001 at 192.168.118.178] has been answered
2010-03-28 21:40:56.510922 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
2010-03-28 21:40:56.530552 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9
2010-03-28 21:40:56.551849 [ERR] switch_rtp.c:2848 Error: zRTP protection drop with code 9
2010-03-28 21:40:56.619175 [ERR] switch_rtp.c:2122 Error: zRTP protection drop with code 9
 
 




 


 		 	   		  
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From serget68 at msn.com  Wed Mar 31 23:14:38 2010
From: serget68 at msn.com (SERGE TUMBA)
Date: Thu, 1 Apr 2010 01:14:38 -0500
Subject: [Freeswitch-users] ZRTP protocol measurement
Message-ID: <COL108-W11B3E7D9EEF7FA21F093D0D91D0@phx.gbl>


I would like to know how to measure the performence of the zrtp using FreeSWITCH that connect two X-Lite softphones which use the zfone for encrypting voice packets on both end phones.

 

Also, can someone contrast and compare ZRTP to SRTP focusing on these two protocol behaviors.

 

Thank you!




 


 		 	   		  
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